Поиск не помог.
Есть Asterisk 16.9.0 и два софтфона, первый Zoiper5 на винде, второй Zoiper на айфон (куплен кодек h264).
Суть проблемы, если звонить с айфона на Zoiper5, то звонок проходит моментально, видео работает.
А вот если звонить наоборот, с винды на айфон звонок проходит только через 3-5 секунды и видео не работает.
Zoiper5 на винде - 12
Zoiper на айфон - 10
sip.conf:
Код: Выделить всё
[general]
context=public
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=tcp,udp
srvlookup=yes
videosupport = always
maxcallbitrate = 2048
directmedia=nonat
encryption=no
[authentication]
[basic-options](!)
dtmfmode=rfc2833
context=from-office
type=friend
[natted-phone](!,basic-options)
directmedia=no
host=dynamic
[public-phone](!,basic-options)
directmedia=yes
[my-codecs](!)
disallow=all
allow=ilbc
allow=g729
allow=gsm
allow=g723
allow=ulaw
[ulaw-phone](!)
disallow=all
allow=ulaw
[10]
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0
type=friend
secret=9999
qualify=yes
port=5060
pickupgroup=1
nat=no
context=test
mailbox=10@device
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=h264
dial=SIP/10
canreinvite=no
callgroup=1
callerid=Test1 <10>
call-limit=2
[12]
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0
type=friend
secret=9999
qualify=yes
port=5060
pickupgroup=1
nat=no
context=test
mailbox=12@device
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=h264
dial=SIP/12
canreinvite=no
callgroup=1
callerid=Test3 <12>
call-limit=2
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
SIP Debugging enabled
<--- SIP read from UDP:10.0.10.99:39918 --->
SUBSCRIBE sip:asterisk@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---afb83b457468eaa4;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>;tag=as130d6b81
From: <sip:12@10.0.0.4;transport=UDP>;tag=6132a955
Call-ID: WfyHO5ID1rHSFL3Awh7w6g..
CSeq: 5 SUBSCRIBE
Expires: 60
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="3da87c50",uri="sip:asterisk@10.0.0.4:5060",response="365f90e7c589b402e984a5c9d556d592",algorithm=MD5
Event: message-summary
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (16 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---afb83b457468eaa4;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=6132a955
To: <sip:12@10.0.0.4;transport=UDP>;tag=as130d6b81
Call-ID: WfyHO5ID1rHSFL3Awh7w6g..
CSeq: 5 SUBSCRIBE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'WfyHO5ID1rHSFL3Awh7w6g..' in 32000 ms (Method: SUBSCRIBE)
<--- SIP read from UDP:10.0.10.99:39918 --->
REGISTER sip:10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---db6e39b36d544d70;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 57 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="62bd0023",uri="sip:10.0.0.4;transport=UDP",response="63f201018282574c778ee1f537355d54",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Sending to 10.0.10.99:39918 (no NAT)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---db6e39b36d544d70;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
To: <sip:12@10.0.0.4;transport=UDP>;tag=as1d11f17f
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 57 REGISTER
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="439632b6"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'gx5dTKfcJejErUc1HNlrhQ..' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.0.10.99:39918 --->
REGISTER sip:10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---399302c4c66aaf69;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 58 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="439632b6",uri="sip:10.0.0.4;transport=UDP",response="05cc8cda9a4b4fa4c1c88e25c777c6bb",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Reliably Transmitting (no NAT) to 10.0.10.99
OPTIONS sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2aca906c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as228a1738
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
Contact: <sip:asterisk@10.0.0.4:5060>
Call-ID: 33c870ec6877a69002bb0d134a9524c7@10.0.0.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:12:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---399302c4c66aaf69;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
To: <sip:12@10.0.0.4;transport=UDP>;tag=as1d11f17f
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 58 REGISTER
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;expires=60
Date: Thu, 23 Apr 2020 16:12:29 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '609f3ac5298c03cd4bd6efb2230cc85f@10.0.0.4:5060' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 10.0.10.99
NOTIFY sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK5403d70f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as0080dc60
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
Contact: <sip:asterisk@10.0.0.4:5060>
Call-ID: 609f3ac5298c03cd4bd6efb2230cc85f@10.0.0.4:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 16.9.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:asterisk@10.0.0.4
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog 'gx5dTKfcJejErUc1HNlrhQ..' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2aca906c
Contact: <sip:10.0.10.99:39918>
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=80100a0c
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as228a1738
Call-ID: 33c870ec6877a69002bb0d134a9524c7@10.0.0.4:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.3.8 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '33c870ec6877a69002bb0d134a9524c7@10.0.0.4:5060' Method: OPTIONS
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK5403d70f
Contact: <sip:10.0.10.99:39918>
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=ba0a5053
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as0080dc60
Call-ID: 609f3ac5298c03cd4bd6efb2230cc85f@10.0.0.4:5060
CSeq: 102 NOTIFY
User-Agent: Z 5.3.8 rv2.9.30
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '609f3ac5298c03cd4bd6efb2230cc85f@10.0.0.4:5060' Method: NOTIFY
<--- SIP read from UDP:10.0.10.99:39918 --->
INVITE sip:10@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---0504a5c1271901b8;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:10@10.0.0.4>
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.3.8 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 501
v=0
o=Z 1587658352901 1 IN IP4 194.15.118.234
s=Z
c=IN IP4 194.15.118.234
t=0 0
m=audio 8000 RTP/AVP 106 9 98 101 0 8 18 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 maxplaybackrate=16000; sprop-maxcapturerate=16000; minptime=20; cbr=1; maxaveragebitrate=20000; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
m=video 8002 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------->
--- (13 headers 18 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Sending to 10.0.10.99:39918 (no NAT)
Using INVITE request as basis request - rB6IlMVd-vKoUpr2vN0Y8Q..
Found peer '12' for '12' from 10.0.10.99:39918
<--- Reliably Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---0504a5c1271901b8;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
To: <sip:10@10.0.0.4>;tag=as55a0cbb5
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 1 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ab5e737"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'rB6IlMVd-vKoUpr2vN0Y8Q..' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.0.10.99:39918 --->
ACK sip:10@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---0504a5c1271901b8;rport
Max-Forwards: 70
To: <sip:10@10.0.0.4>;tag=as55a0cbb5
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:39918 --->
INVITE sip:10@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---3a995de4d64eedc4;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:10@10.0.0.4>
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="1ab5e737",uri="sip:10@10.0.0.4;transport=UDP",response="d9097953daaf2fd690d92687f3475ff3",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 501
v=0
o=Z 1587658352901 1 IN IP4 194.15.118.234
s=Z
c=IN IP4 194.15.118.234
t=0 0
m=audio 8000 RTP/AVP 106 9 98 101 0 8 18 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 maxplaybackrate=16000; sprop-maxcapturerate=16000; minptime=20; cbr=1; maxaveragebitrate=20000; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
m=video 8002 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Using INVITE request as basis request - rB6IlMVd-vKoUpr2vN0Y8Q..
Found peer '12' for '12' from 10.0.10.99:39918
Got SDP version 1 and unique parts [Z 1587658352901 IN IP4 194.15.118.234]
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 98
Found RTP audio format 101
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 3
Found audio description format opus for ID 106
Found unknown media description format telephone-event for ID 98
Found audio description format telephone-event for ID 101
Found audio description format G729 for ID 18
Found RTP video format 118
Found video description format H264 for ID 118
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw|gsm|alaw|g722|g729|opus)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 194.15.118.234:8000
Peer video RTP is at port 194.15.118.234:8002
Looking for 10 in test (domain 10.0.0.4)
sip_route_dump: route/path hop: <sip:12@10.0.10.99:39918;transport=UDP>
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---3a995de4d64eedc4;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
To: <sip:10@10.0.0.4>
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:10@10.0.0.4:5060>
Content-Length: 0
<------------>
Audio is at 16962
Video is at 10.0.0.4:19938
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.10.99:5061:
INVITE sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7600df90
Max-Forwards: 70
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>
Contact: <sip:12@10.0.0.4:5060>
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:12:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326
v=0
o=root 322208777 322208777 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 16962 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 19938 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
---
Retransmitting #1 (no NAT) to 10.0.10.99:5061:
INVITE sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7600df90
Max-Forwards: 70
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>
Contact: <sip:12@10.0.0.4:5060>
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:12:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326
v=0
o=root 322208777 322208777 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 16962 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 19938 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
---
<--- SIP read from UDP:10.0.10.99:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7600df90
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7600df90
Contact: <sip:10@10.0.10.99:5061>
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rv2.9.27
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:10@10.0.10.99:5061>
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---3a995de4d64eedc4;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
To: <sip:10@10.0.0.4>;tag=as5f05e4dd
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:10@10.0.0.4:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:10.0.10.99:39918 --->
<------------->
<--- SIP read from UDP:10.0.10.99:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7600df90
Contact: <sip:10@10.0.10.99:5061>
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.9.27
Allow-Events: presence, kpml, talk
Content-Length: 293
v=0
o=Z 0 3 IN IP4 10.0.10.99
s=Z
c=IN IP4 10.0.10.99
t=0 0
m=audio 8000 RTP/AVP 0 3 110 97 8 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
m=video 0 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
<------------->
--- (12 headers 15 lines) ---
Got SDP version 3 and unique parts [Z 0 IN IP4 10.0.10.99]
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.10.99:8000
Peer doesn't provide video
sip_route_dump: route/path hop: <sip:10@10.0.10.99:5061>
set_destination: Parsing <sip:10@10.0.10.99:5061> for address/port to send to
set_destination: set destination to 10.0.10.99:5061
Transmitting (no NAT) to 10.0.10.99:5061:
ACK sip:10@10.0.10.99:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK347bf4c7
Max-Forwards: 70
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
Contact: <sip:12@10.0.0.4:5060>
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.9.0
Content-Length: 0
---
Audio is at 15900
Video is at 10.0.0.4:13402
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---3a995de4d64eedc4;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
To: <sip:10@10.0.0.4>;tag=as5f05e4dd
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:10@10.0.0.4:5060>
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 504610902 504610902 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 15900 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 13402 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------>
<--- SIP read from UDP:10.0.10.99:39918 --->
ACK sip:10@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---4f962772f1ffdc9a;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:10@10.0.0.4>;tag=as5f05e4dd
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 2 ACK
User-Agent: Z 5.3.8 rv2.9.30
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:5061 --->
INVITE sip:12@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---4145c41218ce6e7f;rport
Max-Forwards: 70
Contact: <sip:10@10.0.10.99:5061>
To: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
From: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.9.27
Allow-Events: presence, kpml, talk
Content-Length: 296
v=0
o=Z 0 4 IN IP4 10.0.10.99
s=Z
c=IN IP4 10.0.10.99
t=0 0
m=audio 8000 RTP/AVP 0 3 110 97 8 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
m=video 8012 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 10.0.10.99:5061 (no NAT)
Comparing SDP version 3 -> 4 and unique parts [Z 0 IN IP4 10.0.10.99] -> [Z 0 IN IP4 10.0.10.99]
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Found RTP video format 99
Found video description format H264 for ID 99
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.10.99:8000
Peer video RTP is at port 10.0.10.99:8012
<--- Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---4145c41218ce6e7f;received=10.0.10.99;rport=5061
From: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
To: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:12@10.0.0.4:5060>
Content-Length: 0
<------------>
Audio is at 16962
Video is at 10.0.0.4:19938
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---4145c41218ce6e7f;received=10.0.10.99;rport=5061
From: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
To: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:12@10.0.0.4:5060>
Content-Type: application/sdp
Content-Length: 326
v=0
o=root 322208777 322208778 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 16962 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 19938 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
<------------>
<--- SIP read from UDP:10.0.10.99:5061 --->
ACK sip:12@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---9099bbb2462ffbc1;rport
Max-Forwards: 70
Contact: <sip:10@10.0.10.99:5061>
To: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
From: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 2 ACK
User-Agent: Zoiper rv2.9.27
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:39918 --->
SUBSCRIBE sip:12@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---341860ef9ef7bdcb;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=0e720940
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 1 SUBSCRIBE
Expires: 60
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Event: message-summary
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (15 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Creating new subscription
Sending to 10.0.10.99:39918 (no NAT)
sip_route_dump: route/path hop: <sip:12@10.0.10.99:39918;transport=UDP>
Found peer '12' for '12' from 10.0.10.99:39918
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---341860ef9ef7bdcb;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=0e720940
To: <sip:12@10.0.0.4;transport=UDP>;tag=as7d5d6061
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54f63155"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'sUoktASYgAJb90A4TTmoEw..' in 6400 ms (Method: SUBSCRIBE)
<--- SIP read from UDP:10.0.10.99:39918 --->
SUBSCRIBE sip:12@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---f0ce259b3f1daca2;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=0e720940
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 2 SUBSCRIBE
Expires: 60
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="54f63155",uri="sip:12@10.0.0.4;transport=UDP",response="5335c9fabf4bd797da738c4894b01466",algorithm=MD5
Event: message-summary
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (16 headers 0 lines) ---
Creating new subscription
Sending to 10.0.10.99:39918 (no NAT)
Found peer '12' for '12' from 10.0.10.99:39918
Scheduling destruction of SIP dialog 'sUoktASYgAJb90A4TTmoEw..' in 70000 ms (Method: SUBSCRIBE)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---f0ce259b3f1daca2;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=0e720940
To: <sip:12@10.0.0.4;transport=UDP>;tag=as7d5d6061
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:12@10.0.0.4:5060>;expires=60
Content-Length: 0
<------------>
Reliably Transmitting (no NAT) to 10.0.10.99
NOTIFY sip:12@10.0.10.99:39918;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK043804df;rport
Max-Forwards: 70
Route: <sip:12@10.0.10.99:39918;transport=UDP>
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as7d5d6061
To: <sip:12@10.0.10.99:39918;transport=UDP>;tag=0e720940
Contact: <sip:asterisk@10.0.0.4:5060>
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 16.9.0
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:asterisk@10.0.0.4
Voice-Message: 0/0 (0/0)
---
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK043804df;rport=5060
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:12@10.0.10.99:39918;transport=UDP>;tag=0e720940
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as7d5d6061
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 102 NOTIFY
User-Agent: Z 5.3.8 rv2.9.30
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (no NAT) to 10.0.10.99:60135:
OPTIONS sip:17@10.0.10.99:60135;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.0.0.4:5060;branch=z9hG4bK1f56fa1b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as6953c13b
To: <sip:17@10.0.10.99:60135;transport=tcp>
Contact: <sip:asterisk@10.0.0.4:5060;transport=tcp>
Call-ID: 4d3557243d744fec0490a33d52565980@10.0.0.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:12:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from TCP:10.0.10.99:60135 --->
SIP/2.0 200 OK
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as6953c13b
To: <sip:17@10.0.10.99:60135;transport=tcp>
Call-ID: 4d3557243d744fec0490a33d52565980@10.0.0.4:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/TCP 10.0.0.4:5060;branch=z9hG4bK1f56fa1b
Supported: eventlist,outbound,replaces
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,MESSAGE,REFER,INFO,PUBLISH,UPDATE
User-Agent: Avaya Vantage Connect/2.2.0.2 (0003; K175D02A; 316.1.41)
Content-Type: application/sdp
Content-Length: 911
v=0
o= 0 1 IN IP4 192.168.88.105
s=-
c=IN IP4 192.168.88.105
b=TIAS:576000
t=0 0
a=activetalker:1
m=audio 0 RTP/AVP 116 9 8 0 110 18 120
c=IN IP4 0.0.0.0
a=ptime:20
a=rtpmap:116 OPUS/48000/2
a=fmtp:116 maxplaybackrate=16000; sprop-maxcapturerate=16000; maxaveragebitrate=20000; stereo=0; sprop-stereo=0; cbr=0; useinbandfec=1; usedtx=0
a=minptime:20
a=maxptime:20
a=rtpmap:9 G722/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:110 G726-32/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:120 telephone-event/8000
m=video 0 RTP/AVP 100 98 113 114
c=IN IP4 0.0.0.0
b=TIAS:512000
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=640c1e;max-mbps=27600;max-fs=920;packetization-mode=1
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=42401e;max-mbps=27600;max-fs=920;packetization-mode=0
a=rtpmap:113 rtx/90000
a=rtpmap:114 rtx/90000
a=rtcp-fb:* ccm fir
a=rtcp-fb:* nack pli
<------------->
--- (11 headers 31 lines) ---
Really destroying SIP dialog '4d3557243d744fec0490a33d52565980@10.0.0.4:5060' Method: OPTIONS
<--- SIP read from UDP:10.0.10.99:5061 --->
<------------->
Really destroying SIP dialog 'N0OhLN8r_mzawF4Do7arKw..' Method: REGISTER
<--- SIP read from UDP:10.0.10.99:39918 --->
BYE sip:10@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---7de91467e210807c;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:10@10.0.0.4>;tag=as5f05e4dd
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 3 BYE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="1ab5e737",uri="sip:10@10.0.0.4:5060",response="bfcefeeba14afeff69d0d9448ba2adba",algorithm=MD5
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Scheduling destruction of SIP dialog 'rB6IlMVd-vKoUpr2vN0Y8Q..' in 6400 ms (Method: BYE)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---7de91467e210807c;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
To: <sip:10@10.0.0.4>;tag=as5f05e4dd
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 3 BYE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:10@10.0.10.99:5061> for address/port to send to
set_destination: set destination to 10.0.10.99:5061
Reliably Transmitting (no NAT) to 10.0.10.99:5061:
BYE sip:10@10.0.10.99:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2ee72871;rport
Max-Forwards: 70
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 16.9.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:10.0.10.99:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2ee72871;rport=5060
Contact: <sip:10@10.0.10.99:5061>
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 103 BYE
User-Agent: Zoiper rv2.9.27
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060' Method: ACK
<--- SIP read from UDP:10.0.10.99:39918 --->
SUBSCRIBE sip:asterisk@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---afb83b457468eaa4;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>;tag=as130d6b81
From: <sip:12@10.0.0.4;transport=UDP>;tag=6132a955
Call-ID: WfyHO5ID1rHSFL3Awh7w6g..
CSeq: 5 SUBSCRIBE
Expires: 60
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="3da87c50",uri="sip:asterisk@10.0.0.4:5060",response="365f90e7c589b402e984a5c9d556d592",algorithm=MD5
Event: message-summary
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (16 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---afb83b457468eaa4;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=6132a955
To: <sip:12@10.0.0.4;transport=UDP>;tag=as130d6b81
Call-ID: WfyHO5ID1rHSFL3Awh7w6g..
CSeq: 5 SUBSCRIBE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'WfyHO5ID1rHSFL3Awh7w6g..' in 32000 ms (Method: SUBSCRIBE)
<--- SIP read from UDP:10.0.10.99:39918 --->
REGISTER sip:10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---db6e39b36d544d70;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 57 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="62bd0023",uri="sip:10.0.0.4;transport=UDP",response="63f201018282574c778ee1f537355d54",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Sending to 10.0.10.99:39918 (no NAT)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---db6e39b36d544d70;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
To: <sip:12@10.0.0.4;transport=UDP>;tag=as1d11f17f
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 57 REGISTER
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="439632b6"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'gx5dTKfcJejErUc1HNlrhQ..' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.0.10.99:39918 --->
REGISTER sip:10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---399302c4c66aaf69;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 58 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="439632b6",uri="sip:10.0.0.4;transport=UDP",response="05cc8cda9a4b4fa4c1c88e25c777c6bb",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Reliably Transmitting (no NAT) to 10.0.10.99
OPTIONS sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2aca906c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as228a1738
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
Contact: <sip:asterisk@10.0.0.4:5060>
Call-ID: 33c870ec6877a69002bb0d134a9524c7@10.0.0.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:12:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---399302c4c66aaf69;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
To: <sip:12@10.0.0.4;transport=UDP>;tag=as1d11f17f
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 58 REGISTER
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;expires=60
Date: Thu, 23 Apr 2020 16:12:29 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '609f3ac5298c03cd4bd6efb2230cc85f@10.0.0.4:5060' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 10.0.10.99
NOTIFY sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK5403d70f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as0080dc60
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
Contact: <sip:asterisk@10.0.0.4:5060>
Call-ID: 609f3ac5298c03cd4bd6efb2230cc85f@10.0.0.4:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 16.9.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:asterisk@10.0.0.4
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog 'gx5dTKfcJejErUc1HNlrhQ..' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2aca906c
Contact: <sip:10.0.10.99:39918>
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=80100a0c
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as228a1738
Call-ID: 33c870ec6877a69002bb0d134a9524c7@10.0.0.4:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.3.8 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '33c870ec6877a69002bb0d134a9524c7@10.0.0.4:5060' Method: OPTIONS
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK5403d70f
Contact: <sip:10.0.10.99:39918>
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=ba0a5053
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as0080dc60
Call-ID: 609f3ac5298c03cd4bd6efb2230cc85f@10.0.0.4:5060
CSeq: 102 NOTIFY
User-Agent: Z 5.3.8 rv2.9.30
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '609f3ac5298c03cd4bd6efb2230cc85f@10.0.0.4:5060' Method: NOTIFY
<--- SIP read from UDP:10.0.10.99:39918 --->
INVITE sip:10@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---0504a5c1271901b8;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:10@10.0.0.4>
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.3.8 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 501
v=0
o=Z 1587658352901 1 IN IP4 194.15.118.234
s=Z
c=IN IP4 194.15.118.234
t=0 0
m=audio 8000 RTP/AVP 106 9 98 101 0 8 18 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 maxplaybackrate=16000; sprop-maxcapturerate=16000; minptime=20; cbr=1; maxaveragebitrate=20000; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
m=video 8002 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------->
--- (13 headers 18 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Sending to 10.0.10.99:39918 (no NAT)
Using INVITE request as basis request - rB6IlMVd-vKoUpr2vN0Y8Q..
Found peer '12' for '12' from 10.0.10.99:39918
<--- Reliably Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---0504a5c1271901b8;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
To: <sip:10@10.0.0.4>;tag=as55a0cbb5
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 1 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ab5e737"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'rB6IlMVd-vKoUpr2vN0Y8Q..' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.0.10.99:39918 --->
ACK sip:10@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---0504a5c1271901b8;rport
Max-Forwards: 70
To: <sip:10@10.0.0.4>;tag=as55a0cbb5
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:39918 --->
INVITE sip:10@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---3a995de4d64eedc4;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:10@10.0.0.4>
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="1ab5e737",uri="sip:10@10.0.0.4;transport=UDP",response="d9097953daaf2fd690d92687f3475ff3",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 501
v=0
o=Z 1587658352901 1 IN IP4 194.15.118.234
s=Z
c=IN IP4 194.15.118.234
t=0 0
m=audio 8000 RTP/AVP 106 9 98 101 0 8 18 3
a=rtpmap:106 opus/48000/2
a=fmtp:106 maxplaybackrate=16000; sprop-maxcapturerate=16000; minptime=20; cbr=1; maxaveragebitrate=20000; useinbandfec=1
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
m=video 8002 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Using INVITE request as basis request - rB6IlMVd-vKoUpr2vN0Y8Q..
Found peer '12' for '12' from 10.0.10.99:39918
Got SDP version 1 and unique parts [Z 1587658352901 IN IP4 194.15.118.234]
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 98
Found RTP audio format 101
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 3
Found audio description format opus for ID 106
Found unknown media description format telephone-event for ID 98
Found audio description format telephone-event for ID 101
Found audio description format G729 for ID 18
Found RTP video format 118
Found video description format H264 for ID 118
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw|gsm|alaw|g722|g729|opus)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 194.15.118.234:8000
Peer video RTP is at port 194.15.118.234:8002
Looking for 10 in test (domain 10.0.0.4)
sip_route_dump: route/path hop: <sip:12@10.0.10.99:39918;transport=UDP>
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---3a995de4d64eedc4;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
To: <sip:10@10.0.0.4>
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:10@10.0.0.4:5060>
Content-Length: 0
<------------>
Audio is at 16962
Video is at 10.0.0.4:19938
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.10.99:5061:
INVITE sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7600df90
Max-Forwards: 70
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>
Contact: <sip:12@10.0.0.4:5060>
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:12:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326
v=0
o=root 322208777 322208777 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 16962 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 19938 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
---
Retransmitting #1 (no NAT) to 10.0.10.99:5061:
INVITE sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7600df90
Max-Forwards: 70
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>
Contact: <sip:12@10.0.0.4:5060>
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:12:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326
v=0
o=root 322208777 322208777 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 16962 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 19938 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
---
<--- SIP read from UDP:10.0.10.99:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7600df90
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7600df90
Contact: <sip:10@10.0.10.99:5061>
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Zoiper rv2.9.27
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:10@10.0.10.99:5061>
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---3a995de4d64eedc4;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
To: <sip:10@10.0.0.4>;tag=as5f05e4dd
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:10@10.0.0.4:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:10.0.10.99:39918 --->
<------------->
<--- SIP read from UDP:10.0.10.99:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7600df90
Contact: <sip:10@10.0.10.99:5061>
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.9.27
Allow-Events: presence, kpml, talk
Content-Length: 293
v=0
o=Z 0 3 IN IP4 10.0.10.99
s=Z
c=IN IP4 10.0.10.99
t=0 0
m=audio 8000 RTP/AVP 0 3 110 97 8 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
m=video 0 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
<------------->
--- (12 headers 15 lines) ---
Got SDP version 3 and unique parts [Z 0 IN IP4 10.0.10.99]
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.10.99:8000
Peer doesn't provide video
sip_route_dump: route/path hop: <sip:10@10.0.10.99:5061>
set_destination: Parsing <sip:10@10.0.10.99:5061> for address/port to send to
set_destination: set destination to 10.0.10.99:5061
Transmitting (no NAT) to 10.0.10.99:5061:
ACK sip:10@10.0.10.99:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK347bf4c7
Max-Forwards: 70
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
Contact: <sip:12@10.0.0.4:5060>
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.9.0
Content-Length: 0
---
Audio is at 15900
Video is at 10.0.0.4:13402
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---3a995de4d64eedc4;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
To: <sip:10@10.0.0.4>;tag=as5f05e4dd
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:10@10.0.0.4:5060>
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 504610902 504610902 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 15900 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 13402 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------>
<--- SIP read from UDP:10.0.10.99:39918 --->
ACK sip:10@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---4f962772f1ffdc9a;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:10@10.0.0.4>;tag=as5f05e4dd
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 2 ACK
User-Agent: Z 5.3.8 rv2.9.30
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:5061 --->
INVITE sip:12@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---4145c41218ce6e7f;rport
Max-Forwards: 70
Contact: <sip:10@10.0.10.99:5061>
To: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
From: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.9.27
Allow-Events: presence, kpml, talk
Content-Length: 296
v=0
o=Z 0 4 IN IP4 10.0.10.99
s=Z
c=IN IP4 10.0.10.99
t=0 0
m=audio 8000 RTP/AVP 0 3 110 97 8 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
m=video 8012 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 10.0.10.99:5061 (no NAT)
Comparing SDP version 3 -> 4 and unique parts [Z 0 IN IP4 10.0.10.99] -> [Z 0 IN IP4 10.0.10.99]
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 97
Found audio description format telephone-event for ID 101
Found RTP video format 99
Found video description format H264 for ID 99
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.10.99:8000
Peer video RTP is at port 10.0.10.99:8012
<--- Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---4145c41218ce6e7f;received=10.0.10.99;rport=5061
From: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
To: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:12@10.0.0.4:5060>
Content-Length: 0
<------------>
Audio is at 16962
Video is at 10.0.0.4:19938
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---4145c41218ce6e7f;received=10.0.10.99;rport=5061
From: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
To: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:12@10.0.0.4:5060>
Content-Type: application/sdp
Content-Length: 326
v=0
o=root 322208777 322208778 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 16962 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 19938 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
<------------>
<--- SIP read from UDP:10.0.10.99:5061 --->
ACK sip:12@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---9099bbb2462ffbc1;rport
Max-Forwards: 70
Contact: <sip:10@10.0.10.99:5061>
To: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
From: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 2 ACK
User-Agent: Zoiper rv2.9.27
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:39918 --->
SUBSCRIBE sip:12@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---341860ef9ef7bdcb;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=0e720940
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 1 SUBSCRIBE
Expires: 60
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Event: message-summary
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (15 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Creating new subscription
Sending to 10.0.10.99:39918 (no NAT)
sip_route_dump: route/path hop: <sip:12@10.0.10.99:39918;transport=UDP>
Found peer '12' for '12' from 10.0.10.99:39918
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---341860ef9ef7bdcb;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=0e720940
To: <sip:12@10.0.0.4;transport=UDP>;tag=as7d5d6061
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54f63155"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'sUoktASYgAJb90A4TTmoEw..' in 6400 ms (Method: SUBSCRIBE)
<--- SIP read from UDP:10.0.10.99:39918 --->
SUBSCRIBE sip:12@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---f0ce259b3f1daca2;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=0e720940
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 2 SUBSCRIBE
Expires: 60
Accept: application/simple-message-summary
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="54f63155",uri="sip:12@10.0.0.4;transport=UDP",response="5335c9fabf4bd797da738c4894b01466",algorithm=MD5
Event: message-summary
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (16 headers 0 lines) ---
Creating new subscription
Sending to 10.0.10.99:39918 (no NAT)
Found peer '12' for '12' from 10.0.10.99:39918
Scheduling destruction of SIP dialog 'sUoktASYgAJb90A4TTmoEw..' in 70000 ms (Method: SUBSCRIBE)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---f0ce259b3f1daca2;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=0e720940
To: <sip:12@10.0.0.4;transport=UDP>;tag=as7d5d6061
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:12@10.0.0.4:5060>;expires=60
Content-Length: 0
<------------>
Reliably Transmitting (no NAT) to 10.0.10.99
NOTIFY sip:12@10.0.10.99:39918;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK043804df;rport
Max-Forwards: 70
Route: <sip:12@10.0.10.99:39918;transport=UDP>
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as7d5d6061
To: <sip:12@10.0.10.99:39918;transport=UDP>;tag=0e720940
Contact: <sip:asterisk@10.0.0.4:5060>
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 16.9.0
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:asterisk@10.0.0.4
Voice-Message: 0/0 (0/0)
---
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK043804df;rport=5060
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:12@10.0.10.99:39918;transport=UDP>;tag=0e720940
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as7d5d6061
Call-ID: sUoktASYgAJb90A4TTmoEw..
CSeq: 102 NOTIFY
User-Agent: Z 5.3.8 rv2.9.30
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (no NAT) to 10.0.10.99:60135:
OPTIONS sip:17@10.0.10.99:60135;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.0.0.4:5060;branch=z9hG4bK1f56fa1b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as6953c13b
To: <sip:17@10.0.10.99:60135;transport=tcp>
Contact: <sip:asterisk@10.0.0.4:5060;transport=tcp>
Call-ID: 4d3557243d744fec0490a33d52565980@10.0.0.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:12:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from TCP:10.0.10.99:60135 --->
SIP/2.0 200 OK
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as6953c13b
To: <sip:17@10.0.10.99:60135;transport=tcp>
Call-ID: 4d3557243d744fec0490a33d52565980@10.0.0.4:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/TCP 10.0.0.4:5060;branch=z9hG4bK1f56fa1b
Supported: eventlist,outbound,replaces
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,MESSAGE,REFER,INFO,PUBLISH,UPDATE
User-Agent: Avaya Vantage Connect/2.2.0.2 (0003; K175D02A; 316.1.41)
Content-Type: application/sdp
Content-Length: 911
v=0
o= 0 1 IN IP4 192.168.88.105
s=-
c=IN IP4 192.168.88.105
b=TIAS:576000
t=0 0
a=activetalker:1
m=audio 0 RTP/AVP 116 9 8 0 110 18 120
c=IN IP4 0.0.0.0
a=ptime:20
a=rtpmap:116 OPUS/48000/2
a=fmtp:116 maxplaybackrate=16000; sprop-maxcapturerate=16000; maxaveragebitrate=20000; stereo=0; sprop-stereo=0; cbr=0; useinbandfec=1; usedtx=0
a=minptime:20
a=maxptime:20
a=rtpmap:9 G722/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:110 G726-32/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:120 telephone-event/8000
m=video 0 RTP/AVP 100 98 113 114
c=IN IP4 0.0.0.0
b=TIAS:512000
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=640c1e;max-mbps=27600;max-fs=920;packetization-mode=1
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=42401e;max-mbps=27600;max-fs=920;packetization-mode=0
a=rtpmap:113 rtx/90000
a=rtpmap:114 rtx/90000
a=rtcp-fb:* ccm fir
a=rtcp-fb:* nack pli
<------------->
--- (11 headers 31 lines) ---
Really destroying SIP dialog '4d3557243d744fec0490a33d52565980@10.0.0.4:5060' Method: OPTIONS
<--- SIP read from UDP:10.0.10.99:5061 --->
<------------->
Really destroying SIP dialog 'N0OhLN8r_mzawF4Do7arKw..' Method: REGISTER
<--- SIP read from UDP:10.0.10.99:39918 --->
BYE sip:10@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---7de91467e210807c;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:10@10.0.0.4>;tag=as5f05e4dd
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 3 BYE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="1ab5e737",uri="sip:10@10.0.0.4:5060",response="bfcefeeba14afeff69d0d9448ba2adba",algorithm=MD5
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Scheduling destruction of SIP dialog 'rB6IlMVd-vKoUpr2vN0Y8Q..' in 6400 ms (Method: BYE)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---7de91467e210807c;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=955d9009
To: <sip:10@10.0.0.4>;tag=as5f05e4dd
Call-ID: rB6IlMVd-vKoUpr2vN0Y8Q..
CSeq: 3 BYE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:10@10.0.10.99:5061> for address/port to send to
set_destination: set destination to 10.0.10.99:5061
Reliably Transmitting (no NAT) to 10.0.10.99:5061:
BYE sip:10@10.0.10.99:5061 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2ee72871;rport
Max-Forwards: 70
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 16.9.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:10.0.10.99:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2ee72871;rport=5060
Contact: <sip:10@10.0.10.99:5061>
To: <sip:10@10.0.10.99:5061;rinstance=108f096d67aa2197;transport=UDP>;tag=7fb83877
From: "Test3" <sip:12@10.0.0.4>;tag=as3c66981c
Call-ID: 719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060
CSeq: 103 BYE
User-Agent: Zoiper rv2.9.27
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '719a0398241e401c02f6e41e66bfe8f4@10.0.0.4:5060' Method: ACK
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
<--- SIP read from UDP:10.0.10.99:5061 --->
INVITE sip:12@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---96972bbd885a48c2;rport
Max-Forwards: 70
Contact: <sip:10@10.0.10.99:5061;transport=UDP>
To: <sip:12@10.0.0.4>
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.9.27
Allow-Events: presence, kpml, talk
Content-Length: 310
v=0
o=Z 1587658564169 1 IN IP4 10.0.10.99
s=Z
c=IN IP4 10.0.10.99
t=0 0
m=audio 8000 RTP/AVP 3 101 110 97 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=sendrecv
m=video 8012 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 10.0.10.99:5061 (no NAT)
Sending to 10.0.10.99:5061 (no NAT)
Using INVITE request as basis request - GpMSwck0Vg2lkQ6hgUQK-A..
Found peer '10' for '10' from 10.0.10.99:5061
<--- Reliably Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---96972bbd885a48c2;received=10.0.10.99;rport=5061
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
To: <sip:12@10.0.0.4>;tag=as03931ea2
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 1 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54dca72d"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'GpMSwck0Vg2lkQ6hgUQK-A..' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.0.10.99:5061 --->
ACK sip:12@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---96972bbd885a48c2;rport
Max-Forwards: 70
To: <sip:12@10.0.0.4>;tag=as03931ea2
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:5061 --->
INVITE sip:12@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---694fbf93406d8f89;rport
Max-Forwards: 70
Contact: <sip:10@10.0.10.99:5061;transport=UDP>
To: <sip:12@10.0.0.4>
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.9.27
Authorization: Digest username="10",realm="asterisk",nonce="54dca72d",uri="sip:12@10.0.0.4;transport=UDP",response="953faf951fb8c3db267548cdda806d0b",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 310
v=0
o=Z 1587658564169 1 IN IP4 10.0.10.99
s=Z
c=IN IP4 10.0.10.99
t=0 0
m=audio 8000 RTP/AVP 3 101 110 97 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=sendrecv
m=video 8012 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------->
--- (14 headers 15 lines) ---
Sending to 10.0.10.99:5061 (no NAT)
Using INVITE request as basis request - GpMSwck0Vg2lkQ6hgUQK-A..
Found peer '10' for '10' from 10.0.10.99:5061
Got SDP version 1 and unique parts [Z 1587658564169 IN IP4 10.0.10.99]
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 0
Found audio description format telephone-event for ID 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 97
Found RTP video format 118
Found video description format H264 for ID 118
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.10.99:8000
Peer video RTP is at port 10.0.10.99:8012
Looking for 12 in test (domain 10.0.0.4)
sip_route_dump: route/path hop: <sip:10@10.0.10.99:5061;transport=UDP>
<--- Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---694fbf93406d8f89;received=10.0.10.99;rport=5061
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
To: <sip:12@10.0.0.4>
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:12@10.0.0.4:5060>
Content-Length: 0
<------------>
Audio is at 16336
Video is at 10.0.0.4:13288
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.10.99
INVITE sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK4911f20d
Max-Forwards: 70
From: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
Contact: <sip:10@10.0.0.4:5060>
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:16:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2064968923 2064968923 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 16336 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 13288 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
---
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK4911f20d
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
From: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK4911f20d
Contact: <sip:12@10.0.10.99:39918>
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=01406a4e
From: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:12@10.0.10.99:39918>
<--- Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---694fbf93406d8f89;received=10.0.10.99;rport=5061
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
To: <sip:12@10.0.0.4>;tag=as718c0210
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:12@10.0.0.4:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:10.0.10.99:39918 --->
REGISTER sip:10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---7c481395caac2b74;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 65 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="7c78799e",uri="sip:10.0.0.4;transport=UDP",response="5d531a7e45f73df3847d903a3433ce3a",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Sending to 10.0.10.99:39918 (no NAT)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---7c481395caac2b74;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
To: <sip:12@10.0.0.4;transport=UDP>;tag=as5db33877
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 65 REGISTER
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="606066dd"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'gx5dTKfcJejErUc1HNlrhQ..' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.0.10.99:39918 --->
REGISTER sip:10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---761d42c7d0b539ca;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 66 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="606066dd",uri="sip:10.0.0.4;transport=UDP",response="0829455000d7715eb4d2963b6e0cfc6b",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Reliably Transmitting (no NAT) to 10.0.10.99
OPTIONS sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK681411c1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as7129b89d
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
Contact: <sip:asterisk@10.0.0.4:5060>
Call-ID: 36ed296214e2e2842470e4d3004d412e@10.0.0.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:16:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---761d42c7d0b539ca;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
To: <sip:12@10.0.0.4;transport=UDP>;tag=as5db33877
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 66 REGISTER
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;expires=60
Date: Thu, 23 Apr 2020 16:16:05 GMT
Content-Length: 0
<------------>
Reliably Transmitting (no NAT) to 10.0.10.99
NOTIFY sip:12@10.0.10.99:39918;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2a128ef5;rport
Max-Forwards: 70
Route: <sip:12@10.0.10.99:39918;transport=UDP>
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as36fa30be
To: <sip:12@10.0.10.99:39918;transport=UDP>;tag=3d3ce70a
Contact: <sip:asterisk@10.0.0.4:5060>
Call-ID: ITEV_VvAbAmytbXsciBR4w..
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 16.9.0
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:asterisk@10.0.0.4
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog 'gx5dTKfcJejErUc1HNlrhQ..' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK681411c1
Contact: <sip:10.0.10.99:39918>
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=99793e6e
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as7129b89d
Call-ID: 36ed296214e2e2842470e4d3004d412e@10.0.0.4:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.3.8 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '36ed296214e2e2842470e4d3004d412e@10.0.0.4:5060' Method: OPTIONS
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2a128ef5;rport=5060
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:12@10.0.10.99:39918;transport=UDP>;tag=3d3ce70a
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as36fa30be
Call-ID: ITEV_VvAbAmytbXsciBR4w..
CSeq: 103 NOTIFY
User-Agent: Z 5.3.8 rv2.9.30
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:39918 --->
<------------->
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK4911f20d
Contact: <sip:12@10.0.10.99:39918>
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=01406a4e
From: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.3.8 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 479
v=0
o=Z 0 2 IN IP4 10.0.10.99
s=Z
c=IN IP4 10.0.10.99
t=0 0
m=audio 8002 RTP/AVP 0 106 9 8 18 3 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 maxplaybackrate=16000; sprop-maxcapturerate=16000; minptime=20; cbr=1; maxaveragebitrate=20000; useinbandfec=1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
m=video 8004 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
<------------->
--- (12 headers 18 lines) ---
Got SDP version 2 and unique parts [Z 0 IN IP4 10.0.10.99]
Found RTP audio format 0
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 98
Found audio description format opus for ID 106
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 98
Found RTP video format 99
Found video description format H264 for ID 99
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw|gsm|alaw|g722|g729|opus)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.10.99:8002
Peer video RTP is at port 10.0.10.99:8004
sip_route_dump: route/path hop: <sip:12@10.0.10.99:39918>
set_destination: Parsing <sip:12@10.0.10.99:39918> for address/port to send to
set_destination: set destination to 10.0.10.99:39918
Transmitting (no NAT) to 10.0.10.99
ACK sip:12@10.0.10.99:39918 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7c1da901
Max-Forwards: 70
From: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=01406a4e
Contact: <sip:10@10.0.0.4:5060>
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.9.0
Content-Length: 0
---
Audio is at 18720
Video is at 10.0.0.4:17700
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---694fbf93406d8f89;received=10.0.10.99;rport=5061
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
To: <sip:12@10.0.0.4>;tag=as718c0210
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:12@10.0.0.4:5060>
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 993549518 993549518 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 18720 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 17700 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------>
<--- SIP read from UDP:10.0.10.99:5061 --->
ACK sip:12@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---a81ddafb4f2cf6c0;rport
Max-Forwards: 70
Contact: <sip:10@10.0.10.99:5061;transport=UDP>
To: <sip:12@10.0.0.4>;tag=as718c0210
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 2 ACK
User-Agent: Zoiper rv2.9.27
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from TCP:10.0.10.99:60135 --->
<------------->
<--- SIP read from UDP:10.0.10.99:39918 --->
BYE sip:10@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---6fa1fbdfe45c82ff;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918>
To: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
From: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=01406a4e
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 2 BYE
User-Agent: Z 5.3.8 rv2.9.30
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Scheduling destruction of SIP dialog '6f1417560a33c8be19382a181ee58208@10.0.0.4:5060' in 6400 ms (Method: BYE)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---6fa1fbdfe45c82ff;received=10.0.10.99;rport=39918
From: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=01406a4e
To: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 2 BYE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'GpMSwck0Vg2lkQ6hgUQK-A..' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:10@10.0.10.99:5061;transport=UDP> for address/port to send to
set_destination: set destination to 10.0.10.99:5061
Reliably Transmitting (no NAT) to 10.0.10.99:5061:
BYE sip:10@10.0.10.99:5061;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK5f51a03d;rport
Max-Forwards: 70
From: <sip:12@10.0.0.4>;tag=as718c0210
To: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.9.0
Proxy-Authorization: Digest username="10", realm="asterisk", algorithm=MD5, uri="sip:10.0.0.4", nonce="54dca72d", response="0abb65625a9b7a6c16f26417f1c9e072"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:10.0.10.99:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK5f51a03d;rport=5060
Contact: <sip:10@10.0.10.99:5061;transport=UDP>
To: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
From: <sip:12@10.0.0.4>;tag=as718c0210
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 102 BYE
User-Agent: Zoiper rv2.9.27
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'GpMSwck0Vg2lkQ6hgUQK-A..' Method: ACK
<--- SIP read from UDP:10.0.10.99:5061 --->
INVITE sip:12@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---96972bbd885a48c2;rport
Max-Forwards: 70
Contact: <sip:10@10.0.10.99:5061;transport=UDP>
To: <sip:12@10.0.0.4>
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.9.27
Allow-Events: presence, kpml, talk
Content-Length: 310
v=0
o=Z 1587658564169 1 IN IP4 10.0.10.99
s=Z
c=IN IP4 10.0.10.99
t=0 0
m=audio 8000 RTP/AVP 3 101 110 97 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=sendrecv
m=video 8012 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------->
--- (13 headers 15 lines) ---
Sending to 10.0.10.99:5061 (no NAT)
Sending to 10.0.10.99:5061 (no NAT)
Using INVITE request as basis request - GpMSwck0Vg2lkQ6hgUQK-A..
Found peer '10' for '10' from 10.0.10.99:5061
<--- Reliably Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---96972bbd885a48c2;received=10.0.10.99;rport=5061
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
To: <sip:12@10.0.0.4>;tag=as03931ea2
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 1 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="54dca72d"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'GpMSwck0Vg2lkQ6hgUQK-A..' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.0.10.99:5061 --->
ACK sip:12@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---96972bbd885a48c2;rport
Max-Forwards: 70
To: <sip:12@10.0.0.4>;tag=as03931ea2
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:5061 --->
INVITE sip:12@10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---694fbf93406d8f89;rport
Max-Forwards: 70
Contact: <sip:10@10.0.10.99:5061;transport=UDP>
To: <sip:12@10.0.0.4>
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.9.27
Authorization: Digest username="10",realm="asterisk",nonce="54dca72d",uri="sip:12@10.0.0.4;transport=UDP",response="953faf951fb8c3db267548cdda806d0b",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 310
v=0
o=Z 1587658564169 1 IN IP4 10.0.10.99
s=Z
c=IN IP4 10.0.10.99
t=0 0
m=audio 8000 RTP/AVP 3 101 110 97 8 0
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=sendrecv
m=video 8012 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------->
--- (14 headers 15 lines) ---
Sending to 10.0.10.99:5061 (no NAT)
Using INVITE request as basis request - GpMSwck0Vg2lkQ6hgUQK-A..
Found peer '10' for '10' from 10.0.10.99:5061
Got SDP version 1 and unique parts [Z 1587658564169 IN IP4 10.0.10.99]
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 8
Found RTP audio format 0
Found audio description format telephone-event for ID 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 97
Found RTP video format 118
Found video description format H264 for ID 118
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.10.99:8000
Peer video RTP is at port 10.0.10.99:8012
Looking for 12 in test (domain 10.0.0.4)
sip_route_dump: route/path hop: <sip:10@10.0.10.99:5061;transport=UDP>
<--- Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---694fbf93406d8f89;received=10.0.10.99;rport=5061
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
To: <sip:12@10.0.0.4>
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:12@10.0.0.4:5060>
Content-Length: 0
<------------>
Audio is at 16336
Video is at 10.0.0.4:13288
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.10.99
INVITE sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK4911f20d
Max-Forwards: 70
From: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
Contact: <sip:10@10.0.0.4:5060>
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:16:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 2064968923 2064968923 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 16336 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 13288 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
---
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK4911f20d
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
From: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK4911f20d
Contact: <sip:12@10.0.10.99:39918>
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=01406a4e
From: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:12@10.0.10.99:39918>
<--- Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---694fbf93406d8f89;received=10.0.10.99;rport=5061
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
To: <sip:12@10.0.0.4>;tag=as718c0210
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:12@10.0.0.4:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:10.0.10.99:39918 --->
REGISTER sip:10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---7c481395caac2b74;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 65 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="7c78799e",uri="sip:10.0.0.4;transport=UDP",response="5d531a7e45f73df3847d903a3433ce3a",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Sending to 10.0.10.99:39918 (no NAT)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---7c481395caac2b74;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
To: <sip:12@10.0.0.4;transport=UDP>;tag=as5db33877
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 65 REGISTER
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="606066dd"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'gx5dTKfcJejErUc1HNlrhQ..' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.0.10.99:39918 --->
REGISTER sip:10.0.0.4;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---761d42c7d0b539ca;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
To: <sip:12@10.0.0.4;transport=UDP>
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 66 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.3.8 rv2.9.30
Authorization: Digest username="12",realm="asterisk",nonce="606066dd",uri="sip:10.0.0.4;transport=UDP",response="0829455000d7715eb4d2963b6e0cfc6b",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Reliably Transmitting (no NAT) to 10.0.10.99
OPTIONS sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK681411c1
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as7129b89d
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>
Contact: <sip:asterisk@10.0.0.4:5060>
Call-ID: 36ed296214e2e2842470e4d3004d412e@10.0.0.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.9.0
Date: Thu, 23 Apr 2020 16:16:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---761d42c7d0b539ca;received=10.0.10.99;rport=39918
From: <sip:12@10.0.0.4;transport=UDP>;tag=2a55d225
To: <sip:12@10.0.0.4;transport=UDP>;tag=as5db33877
Call-ID: gx5dTKfcJejErUc1HNlrhQ..
CSeq: 66 REGISTER
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;expires=60
Date: Thu, 23 Apr 2020 16:16:05 GMT
Content-Length: 0
<------------>
Reliably Transmitting (no NAT) to 10.0.10.99
NOTIFY sip:12@10.0.10.99:39918;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2a128ef5;rport
Max-Forwards: 70
Route: <sip:12@10.0.10.99:39918;transport=UDP>
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as36fa30be
To: <sip:12@10.0.10.99:39918;transport=UDP>;tag=3d3ce70a
Contact: <sip:asterisk@10.0.0.4:5060>
Call-ID: ITEV_VvAbAmytbXsciBR4w..
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 16.9.0
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 88
Messages-Waiting: no
Message-Account: sip:asterisk@10.0.0.4
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog 'gx5dTKfcJejErUc1HNlrhQ..' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK681411c1
Contact: <sip:10.0.10.99:39918>
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=99793e6e
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as7129b89d
Call-ID: 36ed296214e2e2842470e4d3004d412e@10.0.0.4:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Z 5.3.8 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '36ed296214e2e2842470e4d3004d412e@10.0.0.4:5060' Method: OPTIONS
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK2a128ef5;rport=5060
Contact: <sip:12@10.0.10.99:39918;transport=UDP>
To: <sip:12@10.0.10.99:39918;transport=UDP>;tag=3d3ce70a
From: "asterisk" <sip:asterisk@10.0.0.4>;tag=as36fa30be
Call-ID: ITEV_VvAbAmytbXsciBR4w..
CSeq: 103 NOTIFY
User-Agent: Z 5.3.8 rv2.9.30
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:10.0.10.99:39918 --->
<------------->
<--- SIP read from UDP:10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK4911f20d
Contact: <sip:12@10.0.10.99:39918>
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=01406a4e
From: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.3.8 rv2.9.30
Allow-Events: presence, kpml, talk
Content-Length: 479
v=0
o=Z 0 2 IN IP4 10.0.10.99
s=Z
c=IN IP4 10.0.10.99
t=0 0
m=audio 8002 RTP/AVP 0 106 9 8 18 3 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 maxplaybackrate=16000; sprop-maxcapturerate=16000; minptime=20; cbr=1; maxaveragebitrate=20000; useinbandfec=1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
m=video 8004 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv
<------------->
--- (12 headers 18 lines) ---
Got SDP version 2 and unique parts [Z 0 IN IP4 10.0.10.99]
Found RTP audio format 0
Found RTP audio format 106
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 101
Found RTP audio format 98
Found audio description format opus for ID 106
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 98
Found RTP video format 99
Found video description format H264 for ID 99
Capabilities: us - (ulaw|alaw|h264), peer - audio=(ulaw|gsm|alaw|g722|g729|opus)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.10.99:8002
Peer video RTP is at port 10.0.10.99:8004
sip_route_dump: route/path hop: <sip:12@10.0.10.99:39918>
set_destination: Parsing <sip:12@10.0.10.99:39918> for address/port to send to
set_destination: set destination to 10.0.10.99:39918
Transmitting (no NAT) to 10.0.10.99
ACK sip:12@10.0.10.99:39918 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK7c1da901
Max-Forwards: 70
From: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
To: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=01406a4e
Contact: <sip:10@10.0.0.4:5060>
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.9.0
Content-Length: 0
---
Audio is at 18720
Video is at 10.0.0.4:17700
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.0.10.99:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---694fbf93406d8f89;received=10.0.10.99;rport=5061
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
To: <sip:12@10.0.0.4>;tag=as718c0210
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 2 INVITE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:12@10.0.0.4:5060>
Content-Type: application/sdp
Content-Length: 328
v=0
o=root 993549518 993549518 IN IP4 10.0.0.4
s=Asterisk PBX 16.9.0
c=IN IP4 10.0.0.4
b=CT:2048
t=0 0
m=audio 18720 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 17700 RTP/AVP 118
a=rtpmap:118 H264/90000
a=sendrecv
<------------>
<--- SIP read from UDP:10.0.10.99:5061 --->
ACK sip:12@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:5061;branch=z9hG4bK-524287-1---a81ddafb4f2cf6c0;rport
Max-Forwards: 70
Contact: <sip:10@10.0.10.99:5061;transport=UDP>
To: <sip:12@10.0.0.4>;tag=as718c0210
From: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 2 ACK
User-Agent: Zoiper rv2.9.27
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from TCP:10.0.10.99:60135 --->
<------------->
<--- SIP read from UDP:10.0.10.99:39918 --->
BYE sip:10@10.0.0.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---6fa1fbdfe45c82ff;rport
Max-Forwards: 70
Contact: <sip:12@10.0.10.99:39918>
To: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
From: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=01406a4e
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 2 BYE
User-Agent: Z 5.3.8 rv2.9.30
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 10.0.10.99:39918 (no NAT)
Scheduling destruction of SIP dialog '6f1417560a33c8be19382a181ee58208@10.0.0.4:5060' in 6400 ms (Method: BYE)
<--- Transmitting (no NAT) to 10.0.10.99:39918 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.10.99:39918;branch=z9hG4bK-524287-1---6fa1fbdfe45c82ff;received=10.0.10.99;rport=39918
From: <sip:12@10.0.10.99:39918;rinstance=a0c84c2d930a398f;transport=UDP>;tag=01406a4e
To: "Test1" <sip:10@10.0.0.4>;tag=as497e88ec
Call-ID: 6f1417560a33c8be19382a181ee58208@10.0.0.4:5060
CSeq: 2 BYE
Server: Asterisk PBX 16.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'GpMSwck0Vg2lkQ6hgUQK-A..' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:10@10.0.10.99:5061;transport=UDP> for address/port to send to
set_destination: set destination to 10.0.10.99:5061
Reliably Transmitting (no NAT) to 10.0.10.99:5061:
BYE sip:10@10.0.10.99:5061;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK5f51a03d;rport
Max-Forwards: 70
From: <sip:12@10.0.0.4>;tag=as718c0210
To: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.9.0
Proxy-Authorization: Digest username="10", realm="asterisk", algorithm=MD5, uri="sip:10.0.0.4", nonce="54dca72d", response="0abb65625a9b7a6c16f26417f1c9e072"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:10.0.10.99:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK5f51a03d;rport=5060
Contact: <sip:10@10.0.10.99:5061;transport=UDP>
To: "10"<sip:10@10.0.0.4;transport=UDP>;tag=62b73f09
From: <sip:12@10.0.0.4>;tag=as718c0210
Call-ID: GpMSwck0Vg2lkQ6hgUQK-A..
CSeq: 102 BYE
User-Agent: Zoiper rv2.9.27
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'GpMSwck0Vg2lkQ6hgUQK-A..' Method: ACK
<--- SIP read from UDP:10.0.10.99:5061 --->
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.10.99:8000
Peer doesn't provide video
В чем может быть причина?
