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Yealink VP530 проблема с видео

Вопросы по использованию и настройке IP телефонов, шлюзов и всего прочего

Модераторы: april22, Zavr2008

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Exploizer
Сообщения: 10
Зарегистрирован: 22 апр 2015, 10:02

Re: Yealink VP530 проблема с видео

Сообщение Exploizer »

Да, как-то о таком варианте я не подумал. Так, конечно, варнинги всякие сыпятся, но это ерунда.
Итак, лог звонка с неработающим видео:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
Audio is at 17906
Video is at 172.20.0.1:17440
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100012 (g722) to SDP
Adding video codec 200004 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.20.0.235:49447:
INVITE sip:105@172.20.0.235:49447;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK300ef7a2;rport
Max-Forwards: 70
From: "Соколов Сергей" <sip:888@172.20.0.1>;tag=as4cfc5f7a
To: <sip:105@172.20.0.235:49447;transport=tcp>
Contact: <sip:888@172.20.0.1:5060;transport=TCP>
Call-ID: 2292745308bf8127121697567d2fe8c2@172.20.0.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Wed, 22 Apr 2015 13:35:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 483

v=0
o=root 1636433024 1636433024 IN IP4 172.20.0.1
s=Asterisk PBX 11.9.0
c=IN IP4 172.20.0.1
b=CT:10000000
t=0 0
m=audio 17906 RTP/AVP 0 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 17440 RTP/AVP 99
b=TIAS:10000000
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=4280D
a=fmtp:99 profile-level-id=4280D
a=imageattr:99 recv [x=640,y=480,q=0.50]
a=sendrecv

---

<--- SIP read from TCP:172.20.0.235:49447 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK300ef7a2;rport
From: "Соколов Сергей" <sip:888@172.20.0.1>;tag=as4cfc5f7a
To: <sip:105@172.20.0.235:49447;transport=tcp>
Call-ID: 2292745308bf8127121697567d2fe8c2@172.20.0.1:5060
Date: Wed, 22 Apr 2015 13:35:23 GMT
CSeq: 102 INVITE
Server: Cisco-CP9971/9.4.2
Contact: <sip:105@172.20.0.235:49447;transport=tcp>;video
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.0.1
Allow-Events: kpml,dialog
Content-Length: 0
Recv-Info: conference
Recv-Info: x-cisco-conference


<------------->
--- (15 headers 0 lines) ---

<--- SIP read from TCP:172.20.0.235:49447 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK300ef7a2;rport
From: "Соколов Сергей" <sip:888@172.20.0.1>;tag=as4cfc5f7a
To: <sip:105@172.20.0.235:49447;transport=tcp>;tag=c4143c96d10d544313f832a9-21b96836
Call-ID: 2292745308bf8127121697567d2fe8c2@172.20.0.1:5060
Date: Wed, 22 Apr 2015 13:35:23 GMT
CSeq: 102 INVITE
Server: Cisco-CP9971/9.4.2
Contact: <sip:105@172.20.0.235:49447;transport=tcp>;video
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.0.1
Allow-Events: kpml,dialog
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
list_route: hop: <sip:105@172.20.0.235:49447;transport=tcp>

<--- SIP read from TCP:172.20.0.235:49447 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK300ef7a2;rport
From: "Соколов Сергей" <sip:888@172.20.0.1>;tag=as4cfc5f7a
To: <sip:105@172.20.0.235:49447;transport=tcp>;tag=c4143c96d10d544313f832a9-21b96836
Call-ID: 2292745308bf8127121697567d2fe8c2@172.20.0.1:5060
Date: Wed, 22 Apr 2015 13:35:24 GMT
CSeq: 102 INVITE
Server: Cisco-CP9971/9.4.2
Contact: <sip:105@172.20.0.235:49447;transport=tcp>;video
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.0.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 244
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 4958 0 IN IP4 172.20.0.235
s=SIP Call
t=0 0
m=audio 11516 RTP/AVP 0 101
c=IN IP4 172.20.0.235
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 0 RTP/AVP 99
c=IN IP4 0.0.0.0

<------------->
--- (17 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
[2015-04-22 16:35:28] WARNING[2321][C-00055f49]: chan_sip.c:10237 process_sdp: Ignoring video stream offer because port number is zero
Capabilities: us - (ulaw|alaw|g722|h264), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.20.0.235:11516
Peer doesn't provide video
list_route: hop: <sip:105@172.20.0.235:49447;transport=tcp>
set_destination: Parsing <sip:105@172.20.0.235:49447;transport=tcp> for address/port to send to
set_destination: set destination to 172.20.0.235:49447
Transmitting (NAT) to 172.20.0.235:49447:
ACK sip:105@172.20.0.235:49447;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK4dfb056a;rport
Max-Forwards: 70
From: "Соколов Сергей" <sip:888@172.20.0.1>;tag=as4cfc5f7a
To: <sip:105@172.20.0.235:49447;transport=tcp>;tag=c4143c96d10d544313f832a9-21b96836
Contact: <sip:888@172.20.0.1:5060;transport=TCP>
Call-ID: 2292745308bf8127121697567d2fe8c2@172.20.0.1:5060
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---

<--- SIP read from TCP:172.20.0.235:49447 --->
BYE sip:888@172.20.0.1:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 172.20.0.235:49447;branch=z9hG4bK514f1f02
From: <sip:105@172.20.0.235:49447;transport=tcp>;tag=c4143c96d10d544313f832a9-21b96836
To: "Соколов Сергей" <sip:888@172.20.0.1>;tag=as4cfc5f7a
Call-ID: 2292745308bf8127121697567d2fe8c2@172.20.0.1:5060
Max-Forwards: 70
Date: Wed, 22 Apr 2015 13:35:26 GMT
CSeq: 101 BYE
User-Agent: Cisco-CP9971/9.4.2
Content-Length: 0
Authorization: Digest username="105",realm="asterisk",uri="sip:888@172.20.0.1:5060;transport=tcp",response="c0e776a0d397c5af4866a25b1a825f15",nonce="63a69fcf",algorithm=MD5


<------------->
--- (11 headers 0 lines) ---
Sending to 172.20.0.235:49447 (NAT)
Scheduling destruction of SIP dialog '2292745308bf8127121697567d2fe8c2@172.20.0.1:5060' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to 172.20.0.235:49447 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.20.0.235:49447;branch=z9hG4bK514f1f02;received=172.20.0.235;rport=49447
From: <sip:105@172.20.0.235:49447;transport=tcp>;tag=c4143c96d10d544313f832a9-21b96836
To: "Соколов Сергей" <sip:888@172.20.0.1>;tag=as4cfc5f7a
Call-ID: 2292745308bf8127121697567d2fe8c2@172.20.0.1:5060
CSeq: 101 BYE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
А вот с видео:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
Audio is at 11068
Video is at 172.20.0.1:15202
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.20.1.73:33030:
INVITE sip:888@172.20.1.73:5062;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK1fa55df1;rport
Max-Forwards: 70
From: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
To: <sip:888@172.20.1.73:5062;transport=TCP>
Contact: <sip:105@172.20.0.1:5060;transport=TCP>
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Wed, 22 Apr 2015 13:54:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 553

v=0
o=root 213360629 213360629 IN IP4 172.20.0.1
s=Asterisk PBX 11.9.0
c=IN IP4 172.20.0.1
b=CT:10000000
t=0 0
m=audio 11068 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=ptime:20
a=sendrecv
m=video 15202 RTP/AVP 99
b=TIAS:10000000
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801E;packetization-mode=0;level-asymmetry-allowed=1
a=fmtp:99 profile-level-id=42801E;packetization-mode=0;level-asymmetry-allowed=1
a=imageattr:99 recv [x=640,y=480,q=0.50]
a=sendrecv

---

<--- SIP read from TCP:172.20.1.73:33030 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK1fa55df1;rport
From: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
To: <sip:888@172.20.1.73:5062;transport=TCP>
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 102 INVITE
User-Agent: VP530P 23.70.14.16
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- SIP read from TCP:172.20.1.73:33030 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK1fa55df1;rport
From: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
To: <sip:888@172.20.1.73:5062;transport=TCP>;tag=3237405539
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 102 INVITE
Contact: <sip:888@172.20.1.73:5062;transport=TCP>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: VP530P 23.70.14.16
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:888@172.20.1.73:5062;transport=TCP>

<--- SIP read from TCP:172.20.1.73:33030 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK1fa55df1;rport
From: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
To: <sip:888@172.20.1.73:5062;transport=TCP>;tag=3237405539
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 102 INVITE
Contact: <sip:888@172.20.1.73:5062;transport=TCP>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: VP530P 23.70.14.16
Content-Length: 346

v=0
o=- 20036 20036 IN IP4 172.20.1.73
s=SDP data
c=IN IP4 172.20.1.73
t=0 0
m=audio 11782 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
m=video 11784 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42800D; packetization-mode=0; max-mbps=11880;
a=sendrecv

<------------->
--- (11 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
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)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.20.1.73:11782
Peer video RTP is at port 172.20.1.73:11784
list_route: hop: <sip:888@172.20.1.73:5062;transport=TCP>
set_destination: Parsing <sip:888@172.20.1.73:5062;transport=TCP> for address/port to send to
set_destination: set destination to 172.20.1.73:5062
Transmitting (NAT) to 172.20.1.73:33030:
ACK sip:888@172.20.1.73:5062;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK0ec8198d;rport
Max-Forwards: 70
From: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
To: <sip:888@172.20.1.73:5062;transport=TCP>;tag=3237405539
Contact: <sip:105@172.20.0.1:5060;transport=TCP>
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---

<--- SIP read from TCP:172.20.1.73:33030 --->
INFO sip:105@172.20.0.1:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 172.20.1.73:5062;branch=z9hG4bK2756545479
From: <sip:888@172.20.1.73:5062;transport=TCP>;tag=3237405539
To: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 103 INFO
Contact: <sip:888@172.20.1.73:5062;transport=TCP>
Content-Type: application/media_control+xml
Max-Forwards: 70
User-Agent: VP530P 23.70.14.16
Content-Length: 145

<?xml version="1.0" encoding="utf-8"?><media_control><vc_primitive><to_encoder><picture_fast_update/></to_encoder></vc_primitive></media_control>
<------------->
--- (11 headers 1 lines) ---
Receiving INFO!

<--- Transmitting (NAT) to 172.20.1.73:33030 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.20.1.73:5062;branch=z9hG4bK2756545479;received=172.20.1.73;rport=33030
From: <sip:888@172.20.1.73:5062;transport=TCP>;tag=3237405539
To: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 103 INFO
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
set_destination: Parsing <sip:888@172.20.1.73:5062;transport=TCP> for address/port to send to
set_destination: set destination to 172.20.1.73:5062
Reliably Transmitting (NAT) to 172.20.1.73:33030:
INFO sip:888@172.20.1.73:5062;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK3b48c217;rport
Max-Forwards: 70
From: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
To: <sip:888@172.20.1.73:5062;transport=TCP>;tag=3237405539
Contact: <sip:105@172.20.0.1:5060;transport=TCP>
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 103 INFO
User-Agent: Asterisk
Content-Type: application/media_control+xml
Content-Length: 205

<?xml version="1.0" encoding="utf-8" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

---

<--- SIP read from TCP:172.20.1.73:33030 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK3b48c217;rport
From: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
To: <sip:888@172.20.1.73:5062;transport=TCP>;tag=3237405539
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 103 INFO
Contact: <sip:888@172.20.1.73:5062;transport=TCP>
User-Agent: VP530P 23.70.14.16
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---

<--- SIP read from TCP:172.20.1.73:33030 --->
INFO sip:105@172.20.0.1:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 172.20.1.73:5062;branch=z9hG4bK995091428
From: <sip:888@172.20.1.73:5062;transport=TCP>;tag=3237405539
To: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 104 INFO
Contact: <sip:888@172.20.1.73:5062;transport=TCP>
Content-Type: application/media_control+xml
Max-Forwards: 70
User-Agent: VP530P 23.70.14.16
Content-Length: 145

<?xml version="1.0" encoding="utf-8"?><media_control><vc_primitive><to_encoder><picture_fast_update/></to_encoder></vc_primitive></media_control>
<------------->
--- (11 headers 1 lines) ---
Receiving INFO!

<--- Transmitting (NAT) to 172.20.1.73:33030 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.20.1.73:5062;branch=z9hG4bK995091428;received=172.20.1.73;rport=33030
From: <sip:888@172.20.1.73:5062;transport=TCP>;tag=3237405539
To: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 104 INFO
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from TCP:172.20.1.73:33030 --->
BYE sip:105@172.20.0.1:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 172.20.1.73:5062;branch=z9hG4bK1922954142
From: <sip:888@172.20.1.73:5062;transport=TCP>;tag=3237405539
To: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 105 BYE
Contact: <sip:888@172.20.1.73:5062;transport=TCP>
Max-Forwards: 70
User-Agent: VP530P 23.70.14.16
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 172.20.1.73:33030 (NAT)
Scheduling destruction of SIP dialog '46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060' in 7744 ms (Method: BYE)

<--- Transmitting (NAT) to 172.20.1.73:33030 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.20.1.73:5062;branch=z9hG4bK1922954142;received=172.20.1.73;rport=33030
From: <sip:888@172.20.1.73:5062;transport=TCP>;tag=3237405539
To: "Соколов С. Г. - 105" <sip:105@172.20.0.1>;tag=as0243cbe4
Call-ID: 46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060
CSeq: 105 BYE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Reliably Transmitting (NAT) to 172.20.1.73:33030:
OPTIONS sip:888@172.20.1.73:5062;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK417f8a0c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.20.0.1>;tag=as0af62221
To: <sip:888@172.20.1.73:5062;transport=TCP>
Contact: <sip:asterisk@172.20.0.1:5060;transport=TCP>
Call-ID: 1f30291c086f085507244b76701ae6de@172.20.0.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Wed, 22 Apr 2015 13:54:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from TCP:172.20.1.73:33030 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 172.20.0.1:5060;branch=z9hG4bK417f8a0c;rport
From: "asterisk" <sip:asterisk@172.20.0.1>;tag=as0af62221
To: <sip:888@172.20.1.73:5062;transport=TCP>;tag=2746997193
Call-ID: 1f30291c086f085507244b76701ae6de@172.20.0.1:5060
CSeq: 102 OPTIONS
User-Agent: VP530P 23.70.14.16
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1f30291c086f085507244b76701ae6de@172.20.0.1:5060' Method: OPTIONS
Really destroying SIP dialog '46f9fd583ab5b71b686bb19a64371979@172.20.0.1:5060' Method: BYE
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ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: Yealink VP530 проблема с видео

Сообщение ded »

Создайте такую секцию в sip_custom.conf

Код: Выделить всё

[video-codecs](!)
disallow=all
allow=alaw,ulaw,h264
video_fmtp=profile-level-id=42801E\;packetization-mode=0\;level-asymmetry-allowed=1
video_btias=1000000
video_imageattr=recv [x=640,y=480,q=0.50]
ciscounified=yes
dndbusy=yes

[105](video-codecs)
type=friend
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Glukinho
Сообщения: 661
Зарегистрирован: 07 янв 2011, 20:05

Re: Yealink VP530 проблема с видео

Сообщение Glukinho »

ded, откуда инфа про такие параметры? Где почитать?
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ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: Yealink VP530 проблема с видео

Сообщение ded »

Это не я гиперумник, это результат работы нашей компании, в которой, что скрывать? есть умники в лучшем смысле этого слова :)
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Glukinho
Сообщения: 661
Зарегистрирован: 07 янв 2011, 20:05

Re: Yealink VP530 проблема с видео

Сообщение Glukinho »

Почитать-то где?)
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ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: Yealink VP530 проблема с видео

Сообщение ded »

Да нигде! Читать только пакеты SIP invite + SDP.
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Glukinho
Сообщения: 661
Зарегистрирован: 07 янв 2011, 20:05

Re: Yealink VP530 проблема с видео

Сообщение Glukinho »

Не хотите делиться :(
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Exploizer
Сообщения: 10
Зарегистрирован: 22 апр 2015, 10:02

Re: Yealink VP530 проблема с видео

Сообщение Exploizer »

С параметрами ничего не изменилось. Логи, правда, еще не смотрел.

Эхо-тест на VP530 работает отлично - себя вижу. На 9971 - черный экран, как будто камера закрыта.

И еще вот какая штука - когда добавил шаблон [video-codecs] к пиру 105 он начал отваливаться, причем во время звонка. При этом разговор идет.
[2015-04-23 09:50:24] NOTICE[19372] chan_sip.c: Peer '105' is now UNREACHABLE! Last qualify: 5
[2015-04-23 09:50:34] NOTICE[17808] chan_sip.c: Peer '105' is now Reachable. (5ms / 3000ms)

Прописываю параметры без шаблона - все нормально, не отваливается. Это вообще парадокс...
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ded
Сообщения: 15625
Зарегистрирован: 26 авг 2010, 19:00

Re: Yealink VP530 проблема с видео

Сообщение ded »

Exploizer писал(а):С параметрами ничего не изменилось. Логи, правда, еще не смотрел.

Эхо-тест на VP530 работает отлично - себя вижу. На 9971 - черный экран, как будто камера закрыта..
Проблема именно на 9971
Посмотрите sip invite + sdp при эхотесте.

В общем случае: изучить SDP и подставить нужные параметры в шаблон видео кодека.
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MIKS
Сообщения: 80
Зарегистрирован: 12 мар 2014, 13:43

Re: Yealink VP530 проблема с видео

Сообщение MIKS »

С fmtp Вообще очень сыро в chan_sip Я не знаю как в pjsip.
Я когда с RTSP через chan_sip игрался fmtp передавал через костылики
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