WebRTC

Otkrick · Сообщение **Otkrick** » 29 янв 2013, 17:12

Из браузера на внутренний exten звонок проходит, но звука нет. Телефоны работают в штатном режиме. Никто не сталкивался?
Пользовался http://highsecurity.blogspot.ru/2012/12 ... ipml5.html

awsswa · Сообщение **awsswa** » 29 янв 2013, 17:21

а у всех такой косят, работает только отправляя звук на default gataway

Otkrick · Сообщение **Otkrick** » 29 янв 2013, 17:25

А можно про default gateway поподробнее?

awsswa · Сообщение **awsswa** » 29 янв 2013, 17:33

http://www.cyberciti.biz/faq/howto-rhel ... t-gateway/

Otkrick · Сообщение **Otkrick** » 29 янв 2013, 17:38

не-не-нее.. Вопрос в том, что машина с asterisk'ом должна быть шлюзом по умолчанию для клиента?

awsswa · Сообщение **awsswa** » 29 янв 2013, 17:50

Вещаете asterisk наружу ( не за роутер а напрямую в интернет ) - регистрируютесь с наружи и пробуете свой WebCRT. Из локалки звука не будет.
Ради этого эксперимента достаточно одного роутера, сервера asterisk и компьютера включеного в wan порт роутера.

Otkrick · Сообщение **Otkrick** » 29 янв 2013, 18:03

В sip.conf realm-реальный адрес, пробросил все udp'ные порты на машину с астериском, пробросил порты 5060 и 8088. Регистрируется снаружи, звонок проходит, звука нет

awsswa · Сообщение **awsswa** » 29 янв 2013, 18:08

Настало время снятия дампа - и глядеть куда звук идет.
И вроде я говорил - за роутером нельзя, только напрямую в интернет - но вы пробуйте, результат тоже интересен.

Otkrick · Сообщение **Otkrick** » 29 янв 2013, 18:16

Роутер прозрачно проксирует все udp-пакеты, поэтому "за роутером нельзя" может быть неактуально. Заметил еще, что теперь веб-сервер не получает информацию о терминации вызова: на телефоне кладу трубку, на странице пишет по-прежнему "In call". В локалке проблемы не было.

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: 1ая попытка звонка из браузера на внутренний номер

14:25:59.966858 IP (tos 0x0, ttl 55, id 4643, offset 0, flags [+], proto UDP (17), length 1500)
my.sip-provider.com.8060 > asterisk.corp.local.sip: SIP, length: 1472
INVITE sip:105@213.87.91.67 SIP/2.0
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---ee36d90527c6653b;rport
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKzddB8q1MxmmV4NnzsDt1Z0Takwah3bAx;rport=1049;received=213.87.91.67
Max-Forwards: 69
Contact: "Test user"<sip:8000@86.152.30.40:8060;transport=udp;ws-src-ip=213.87.91.67;ws-src-port=1049;rtcweb-breaker=no>;language="en,fr";+g.oma.sip-im;+sip.ice
To: <sip:105@213.87.91.67>
From: <sip:8000@213.87.91.67>;tag=VlUFqURtZ6AgDlaWDVTl
Call-ID: d4319c29-fe76-489e-2d64-bbd18371d536
CSeq: 42650 INVITE
Content-Type: application/sdp
Organization: Doubango Telecom
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
Content-Length: 1293

v=0
o=- 2615996607 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
m=audio 1893 RTP/SAVPF 103 104 111 0 8 106 105 13 126
c=IN IP4 213.87.91.67
a=rtcp:1893 IN IP4 213.87.91.67
a=candidate:1259018718 1 udp 2113937151 192.168.0.101 1893 typ host generation 0
a=candidate:1259018718 1 udp 1677729535 213.87.91.67 1893 typ srflx generation 0
a=candidate:1259018718 2 udp 2113937151 192.168.0.101 1893 typ host generation 0
a=candidate:1259018718 2 udp 1677729535 213.87.91.67 1893 typ srflx generation 0
a=ice-ufrag:kLxSLRjY9+p7X9FP
a=ice-pwd:SAN+xj50bWjRaKfbEYDnvHni
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:9iwflVjm/qmc1Js0S6D3p8G8wrgNyvcvjWNBYzHz
a=[|sip]
14:25:59.967330 IP (tos 0x0, ttl 64, id 1503, offset 0, flags [none], proto UDP (17), length 696)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 668
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---ee36d90527c6653b;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKzddB8q1MxmmV4NnzsDt1Z0Takwah3bAx;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=VlUFqURtZ6AgDlaWDVTl
To: <sip:105@213.87.91.67>;tag=as70b4f0ad
Call-ID: d4319c29-fe76-489e-2d64-bbd18371d536
CSeq: 42650 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="213.87.91.67", nonce="6d63f2ff"
Content-Length: 0

14:26:00.025129 IP (tos 0x0, ttl 55, id 0, offset 0, flags [DF], proto UDP (17), length 353)
my.sip-provider.com.8060 > asterisk.corp.local.sip: SIP, length: 325
ACK sip:105@213.87.91.67 SIP/2.0
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---ee36d90527c6653b;rport
Max-Forwards: 70
To: <sip:105@213.87.91.67>;tag=as70b4f0ad
From: <sip:8000@213.87.91.67>;tag=VlUFqURtZ6AgDlaWDVTl
Call-ID: d4319c29-fe76-489e-2d64-bbd18371d536
CSeq: 42650 ACK
Content-Length: 0

14:26:00.339372 IP (tos 0x0, ttl 55, id 4644, offset 0, flags [+], proto UDP (17), length 1500)
my.sip-provider.com.8060 > asterisk.corp.local.sip: SIP, length: 1472
INVITE sip:105@213.87.91.67 SIP/2.0
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---aeae6c282ff5e672;rport
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKLwGDKXTkxxxaiCwD5XbLTomTDVH0R1TE;rport=1049;received=213.87.91.67
Max-Forwards: 69
Contact: "Test user"<sip:8000@86.152.30.40:8060;transport=udp;ws-src-ip=213.87.91.67;ws-src-port=1049;rtcweb-breaker=no>;language="en,fr";+g.oma.sip-im;+sip.ice
To: <sip:105@213.87.91.67>
From: <sip:8000@213.87.91.67>;tag=VlUFqURtZ6AgDlaWDVTl
Call-ID: d4319c29-fe76-489e-2d64-bbd18371d536
CSeq: 42651 INVITE
Content-Type: application/sdp
Organization: Doubango Telecom
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
Authorization: Digest username="8000",realm="213.87.91.67",nonce="6d63f2ff",uri="sip:105@213.87.91.67",response="7b36f299cf5b8824bab7d8338faefaab",algorithm=MD5
Content-Length: 1293

v=0
o=- 2615996607 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
m=audio 1893 RTP/SAVPF 103 104 111 0 8 106 105 13 126
c=IN IP4 213.87.91.67
a=rtcp:1893 IN IP4 213.87.91.67
a=candidate:1259018718 1 udp 2113937151 192.168.0.101 1893 typ host generation 0
a=candidate:1259018718 1 udp 1677729535 213.87.91.67 1893 typ srflx generation 0
a=candidate:1259018718 2 udp 2113937151 192.168.0.101 1893 typ host generation 0
a=candidate:1259018718 2 udp 1677729535 213.87.91.67 1893 typ srflx generation 0
a=ice-ufrag:kLxSLRjY9+p7X9FP
a=ice-pwd:SAN+xj50bWj[|sip]
14:26:00.377907 IP (tos 0x0, ttl 64, id 1504, offset 0, flags [none], proto UDP (17), length 631)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 603
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---aeae6c282ff5e672;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKLwGDKXTkxxxaiCwD5XbLTomTDVH0R1TE;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=VlUFqURtZ6AgDlaWDVTl
To: <sip:105@213.87.91.67>
Call-ID: d4319c29-fe76-489e-2d64-bbd18371d536
CSeq: 42651 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Length: 0

14:26:00.418144 IP (tos 0x0, ttl 64, id 56939, offset 0, flags [none], proto UDP (17), length 882)
asterisk.corp.local.sip > 192.168.0.128.sip: SIP, length: 854
INVITE sip:105@192.168.0.128:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK6c5cac49
Max-Forwards: 70
From: <sip:8000@192.168.0.9>;tag=as34f6f6ab
To: <sip:105@192.168.0.128:5060>
Contact: <sip:8000@192.168.0.9:5060>
Call-ID: 5c22956125c7d15911284e3828729d7c@192.168.0.9:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.2.1
Date: Tue, 29 Jan 2013 14:26:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 305

v=0
o=root 437821690 437821690 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 17620 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

14:26:00.424271 IP (tos 0x68, ttl 63, id 9146, offset 0, flags [none], proto UDP (17), length 313)
192.168.0.128.sip > asterisk.corp.local.sip: SIP, length: 285
SIP/2.0 100 Trying
To: <sip:105@192.168.0.128:5060>
From: <sip:8000@192.168.0.9>;tag=as34f6f6ab
Call-ID: 5c22956125c7d15911284e3828729d7c@192.168.0.9:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK6c5cac49
Server: Cisco/SPA504G-7.5.4
Content-Length: 0

14:26:00.465148 IP (tos 0x68, ttl 63, id 9147, offset 0, flags [none], proto UDP (17), length 396)
192.168.0.128.sip > asterisk.corp.local.sip: SIP, length: 368
SIP/2.0 180 Ringing
To: <sip:105@192.168.0.128:5060>;tag=7a0e22cce7c44cefi0
From: <sip:8000@192.168.0.9>;tag=as34f6f6ab
Call-ID: 5c22956125c7d15911284e3828729d7c@192.168.0.9:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK6c5cac49
Contact: "Molchanov_Anatoly" <sip:105@192.168.0.128:5060>
Server: Cisco/SPA504G-7.5.4
Content-Length: 0

14:26:00.465394 IP (tos 0x0, ttl 64, id 1505, offset 0, flags [none], proto UDP (17), length 647)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 619
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---aeae6c282ff5e672;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKLwGDKXTkxxxaiCwD5XbLTomTDVH0R1TE;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=VlUFqURtZ6AgDlaWDVTl
To: <sip:105@213.87.91.67>;tag=as44ec1ce7
Call-ID: d4319c29-fe76-489e-2d64-bbd18371d536
CSeq: 42651 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Length: 0

14:26:02.049968 IP (tos 0x68, ttl 63, id 9148, offset 0, flags [none], proto UDP (17), length 724)
192.168.0.128.sip > asterisk.corp.local.sip: SIP, length: 696
SIP/2.0 200 OK
To: <sip:105@192.168.0.128:5060>;tag=7a0e22cce7c44cefi0
From: <sip:8000@192.168.0.9>;tag=as34f6f6ab
Call-ID: 5c22956125c7d15911284e3828729d7c@192.168.0.9:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK6c5cac49
Contact: "Molchanov_Anatoly" <sip:105@192.168.0.128:5060>
Server: Cisco/SPA504G-7.5.4
Content-Length: 208
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 638668 638668 IN IP4 192.168.0.128
s=-
c=IN IP4 192.168.0.128
t=0 0
m=audio 16528 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

14:26:02.050404 IP (tos 0x0, ttl 64, id 56940, offset 0, flags [none], proto UDP (17), length 413)
asterisk.corp.local.sip > 192.168.0.128.sip: SIP, length: 385
ACK sip:105@192.168.0.128:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK6c841739
Max-Forwards: 70
From: <sip:8000@192.168.0.9>;tag=as34f6f6ab
To: <sip:105@192.168.0.128:5060>;tag=7a0e22cce7c44cefi0
Contact: <sip:8000@192.168.0.9:5060>
Call-ID: 5c22956125c7d15911284e3828729d7c@192.168.0.9:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0

14:26:02.050686 IP (tos 0x0, ttl 64, id 1506, offset 0, flags [none], proto UDP (17), length 1093)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1065
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---aeae6c282ff5e672;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKLwGDKXTkxxxaiCwD5XbLTomTDVH0R1TE;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=VlUFqURtZ6AgDlaWDVTl
To: <sip:105@213.87.91.67>;tag=as44ec1ce7
Call-ID: d4319c29-fe76-489e-2d64-bbd18371d536
CSeq: 42651 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 418

v=0
o=root 1024550397 1024550397 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 15730 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:60m2LsFuMuExE0EBEyXiMPFEBMsUhrkQBth+a0tU

14:26:02.550025 IP (tos 0x0, ttl 64, id 1507, offset 0, flags [none], proto UDP (17), length 1093)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1065
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---aeae6c282ff5e672;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKLwGDKXTkxxxaiCwD5XbLTomTDVH0R1TE;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=VlUFqURtZ6AgDlaWDVTl
To: <sip:105@213.87.91.67>;tag=as44ec1ce7
Call-ID: d4319c29-fe76-489e-2d64-bbd18371d536
CSeq: 42651 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 418

v=0
o=root 1024550397 1024550397 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 15730 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:60m2LsFuMuExE0EBEyXiMPFEBMsUhrkQBth+a0tU

14:26:03.550893 IP (tos 0x0, ttl 64, id 1508, offset 0, flags [none], proto UDP (17), length 1093)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1065
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---aeae6c282ff5e672;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKLwGDKXTkxxxaiCwD5XbLTomTDVH0R1TE;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=VlUFqURtZ6AgDlaWDVTl
To: <sip:105@213.87.91.67>;tag=as44ec1ce7
Call-ID: d4319c29-fe76-489e-2d64-bbd18371d536
CSeq: 42651 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 418

v=0
o=root 1024550397 1024550397 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 15730 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:60m2LsFuMuExE0EBEyXiMPFEBMsUhrkQBth+a0tU

14:26:05.090273 IP (tos 0x68, ttl 63, id 9298, offset 0, flags [none], proto UDP (17), length 372)
192.168.0.128.sip > asterisk.corp.local.sip: SIP, length: 344
BYE sip:8000@192.168.0.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.128:5060;branch=z9hG4bK-ae8e5886
From: <sip:105@192.168.0.128>;tag=7a0e22cce7c44cefi0
To: <sip:8000@192.168.0.9>;tag=as34f6f6ab
Call-ID: 5c22956125c7d15911284e3828729d7c@192.168.0.9:5060
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Cisco/SPA504G-7.5.4
Content-Length: 0

14:26:05.090535 IP (tos 0x0, ttl 64, id 56941, offset 0, flags [none], proto UDP (17), length 461)
asterisk.corp.local.sip > 192.168.0.128.sip: SIP, length: 433
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.128:5060;branch=z9hG4bK-ae8e5886;received=192.168.0.128
From: <sip:105@192.168.0.128>;tag=7a0e22cce7c44cefi0
To: <sip:8000@192.168.0.9>;tag=as34f6f6ab
Call-ID: 5c22956125c7d15911284e3828729d7c@192.168.0.9:5060
CSeq: 101 BYE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

14:26:05.551139 IP (tos 0x0, ttl 64, id 1509, offset 0, flags [none], proto UDP (17), length 1093)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1065
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---aeae6c282ff5e672;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKLwGDKXTkxxxaiCwD5XbLTomTDVH0R1TE;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=VlUFqURtZ6AgDlaWDVTl
To: <sip:105@213.87.91.67>;tag=as44ec1ce7
Call-ID: d4319c29-fe76-489e-2d64-bbd18371d536
CSeq: 42651 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 418

v=0
o=root 1024550397 1024550397 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 15730 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:60m2LsFuMuExE0EBEyXiMPFEBMsUhrkQBth+a0tU

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: 2ая попытка звонка из браузера на внутренний номер

14:29:01.751095 IP (tos 0x0, ttl 64, id 1529, offset 0, flags [none], proto UDP (17), length 1091)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1063
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---377b116f1c2ead55;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKg7Mqxu7efdyTC7x0b23uSRrzHUuiPvzR;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=TbymCB77EpJJvrqtLlz8
To: <sip:105@213.87.91.67>;tag=as6f39e0b7
Call-ID: 0a591117-cd17-2c1b-09a9-f87fefe976fa
CSeq: 60448 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 416

v=0
o=root 653978788 653978788 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 18604 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:72SXW3MEuhZt7opeu8/YBFG3LBxMRuxQWhYDAh5O

14:29:02.016530 IP (tos 0x0, ttl 55, id 4647, offset 0, flags [+], proto UDP (17), length 1500)
my.sip-provider.com.8060 > asterisk.corp.local.sip: SIP, length: 1472
INVITE sip:105@213.87.91.67 SIP/2.0
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---f9e8651b00944a0e;rport
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bK5IX06TLurKbfVK8ngptvIeDR8J2EH7WM;rport=1049;received=213.87.91.67
Max-Forwards: 69
Contact: "Test user"<sip:8000@86.152.30.40:8060;transport=udp;ws-src-ip=213.87.91.67;ws-src-port=1049;rtcweb-breaker=no>;language="en,fr";+g.oma.sip-im;+sip.ice
To: <sip:105@213.87.91.67>
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19741 INVITE
Content-Type: application/sdp
Organization: Doubango Telecom
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
Content-Length: 1285

v=0
o=- 4100437251 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
m=audio 2043 RTP/SAVPF 103 104 111 0 8 106 105 13 126
c=IN IP4 213.87.91.67
a=rtcp:2043 IN IP4 213.87.91.67
a=candidate:1259018718 1 udp 2113937151 192.168.0.101 80 typ host generation 0
a=candidate:1259018718 1 udp 1677729535 213.87.91.67 2043 typ srflx generation 0
a=candidate:1259018718 2 udp 2113937151 192.168.0.101 80 typ host generation 0
a=candidate:1259018718 2 udp 1677729535 213.87.91.67 2043 typ srflx generation 0
a=ice-ufrag:ylY7CnpSO2cEG+tl
a=ice-pwd:TreDbkHViBdggAk8X0ovIgSR
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:QSIBnb1XWT4uHLDxKpLJorOmnXEKLsW5EOLAqM/n
a=cryp[|sip]
14:29:02.016892 IP (tos 0x0, ttl 64, id 1530, offset 0, flags [none], proto UDP (17), length 696)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 668
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---f9e8651b00944a0e;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bK5IX06TLurKbfVK8ngptvIeDR8J2EH7WM;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
To: <sip:105@213.87.91.67>;tag=as3c4b0bd3
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19741 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="213.87.91.67", nonce="62abd412"
Content-Length: 0

14:29:02.074707 IP (tos 0x0, ttl 55, id 0, offset 0, flags [DF], proto UDP (17), length 353)
my.sip-provider.com.8060 > asterisk.corp.local.sip: SIP, length: 325
ACK sip:105@213.87.91.67 SIP/2.0
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---f9e8651b00944a0e;rport
Max-Forwards: 70
To: <sip:105@213.87.91.67>;tag=as3c4b0bd3
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19741 ACK
Content-Length: 0

14:29:02.461389 IP (tos 0x0, ttl 55, id 4648, offset 0, flags [+], proto UDP (17), length 1500)
my.sip-provider.com.8060 > asterisk.corp.local.sip: SIP, length: 1472
INVITE sip:105@213.87.91.67 SIP/2.0
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---99b4571e317b5877;rport
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKcdJtSCS7uPLcJfKW1ROOB79WBv9G5iEa;rport=1049;received=213.87.91.67
Max-Forwards: 69
Contact: "Test user"<sip:8000@86.152.30.40:8060;transport=udp;ws-src-ip=213.87.91.67;ws-src-port=1049;rtcweb-breaker=no>;language="en,fr";+g.oma.sip-im;+sip.ice
To: <sip:105@213.87.91.67>
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19742 INVITE
Content-Type: application/sdp
Organization: Doubango Telecom
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.01.14
Authorization: Digest username="8000",realm="213.87.91.67",nonce="62abd412",uri="sip:105@213.87.91.67",response="180988f067ded8d0b61796b702ad861b",algorithm=MD5
Content-Length: 1285

v=0
o=- 4100437251 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
m=audio 2043 RTP/SAVPF 103 104 111 0 8 106 105 13 126
c=IN IP4 213.87.91.67
a=rtcp:2043 IN IP4 213.87.91.67
a=candidate:1259018718 1 udp 2113937151 192.168.0.101 80 typ host generation 0
a=candidate:1259018718 1 udp 1677729535 213.87.91.67 2043 typ srflx generation 0
a=candidate:1259018718 2 udp 2113937151 192.168.0.101 80 typ host generation 0
a=candidate:1259018718 2 udp 1677729535 213.87.91.67 2043 typ srflx generation 0
a=ice-ufrag:ylY7CnpSO2cEG+tl
a=ice-pwd:TreDbkHViBdggAk[|sip]
14:29:02.501649 IP (tos 0x0, ttl 64, id 1531, offset 0, flags [none], proto UDP (17), length 631)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 603
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---99b4571e317b5877;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKcdJtSCS7uPLcJfKW1ROOB79WBv9G5iEa;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
To: <sip:105@213.87.91.67>
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19742 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Length: 0

14:29:02.541726 IP (tos 0x0, ttl 64, id 56945, offset 0, flags [none], proto UDP (17), length 884)
asterisk.corp.local.sip > 192.168.0.128.sip: SIP, length: 856
INVITE sip:105@192.168.0.128:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK004f0620
Max-Forwards: 70
From: <sip:8000@192.168.0.9>;tag=as459babd6
To: <sip:105@192.168.0.128:5060>
Contact: <sip:8000@192.168.0.9:5060>
Call-ID: 5ae31da3489d562a5bc81b4c2f507b3d@192.168.0.9:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.2.1
Date: Tue, 29 Jan 2013 14:29:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 1633385113 1633385113 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 10594 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

14:29:02.553436 IP (tos 0x68, ttl 63, id 9481, offset 0, flags [none], proto UDP (17), length 313)
192.168.0.128.sip > asterisk.corp.local.sip: SIP, length: 285
SIP/2.0 100 Trying
To: <sip:105@192.168.0.128:5060>
From: <sip:8000@192.168.0.9>;tag=as459babd6
Call-ID: 5ae31da3489d562a5bc81b4c2f507b3d@192.168.0.9:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK004f0620
Server: Cisco/SPA504G-7.5.4
Content-Length: 0

14:29:02.588202 IP (tos 0x68, ttl 63, id 9482, offset 0, flags [none], proto UDP (17), length 396)
192.168.0.128.sip > asterisk.corp.local.sip: SIP, length: 368
SIP/2.0 180 Ringing
To: <sip:105@192.168.0.128:5060>;tag=def855099b733efai0
From: <sip:8000@192.168.0.9>;tag=as459babd6
Call-ID: 5ae31da3489d562a5bc81b4c2f507b3d@192.168.0.9:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK004f0620
Contact: "Molchanov_Anatoly" <sip:105@192.168.0.128:5060>
Server: Cisco/SPA504G-7.5.4
Content-Length: 0

14:29:02.588483 IP (tos 0x0, ttl 64, id 1532, offset 0, flags [none], proto UDP (17), length 647)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 619
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---99b4571e317b5877;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKcdJtSCS7uPLcJfKW1ROOB79WBv9G5iEa;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
To: <sip:105@213.87.91.67>;tag=as63421bc7
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19742 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Length: 0

14:29:03.849215 IP (tos 0x68, ttl 63, id 9483, offset 0, flags [none], proto UDP (17), length 724)
192.168.0.128.sip > asterisk.corp.local.sip: SIP, length: 696
SIP/2.0 200 OK
To: <sip:105@192.168.0.128:5060>;tag=def855099b733efai0
From: <sip:8000@192.168.0.9>;tag=as459babd6
Call-ID: 5ae31da3489d562a5bc81b4c2f507b3d@192.168.0.9:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK004f0620
Contact: "Molchanov_Anatoly" <sip:105@192.168.0.128:5060>
Server: Cisco/SPA504G-7.5.4
Content-Length: 208
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 656880 656880 IN IP4 192.168.0.128
s=-
c=IN IP4 192.168.0.128
t=0 0
m=audio 16532 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

14:29:03.849703 IP (tos 0x0, ttl 64, id 56946, offset 0, flags [none], proto UDP (17), length 413)
asterisk.corp.local.sip > 192.168.0.128.sip: SIP, length: 385
ACK sip:105@192.168.0.128:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.9:5060;branch=z9hG4bK492b1fc1
Max-Forwards: 70
From: <sip:8000@192.168.0.9>;tag=as459babd6
To: <sip:105@192.168.0.128:5060>;tag=def855099b733efai0
Contact: <sip:8000@192.168.0.9:5060>
Call-ID: 5ae31da3489d562a5bc81b4c2f507b3d@192.168.0.9:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.2.1
Content-Length: 0

14:29:03.849964 IP (tos 0x0, ttl 64, id 1533, offset 0, flags [none], proto UDP (17), length 1093)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1065
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---99b4571e317b5877;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKcdJtSCS7uPLcJfKW1ROOB79WBv9G5iEa;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
To: <sip:105@213.87.91.67>;tag=as63421bc7
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19742 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 418

v=0
o=root 2117399007 2117399007 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 14962 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:0lKXAKiA8dtcR20k4u2TCLFhftqcDvad1spIhTd4

14:29:04.349169 IP (tos 0x0, ttl 64, id 1534, offset 0, flags [none], proto UDP (17), length 1093)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1065
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---99b4571e317b5877;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKcdJtSCS7uPLcJfKW1ROOB79WBv9G5iEa;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
To: <sip:105@213.87.91.67>;tag=as63421bc7
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19742 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 418

v=0
o=root 2117399007 2117399007 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 14962 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:0lKXAKiA8dtcR20k4u2TCLFhftqcDvad1spIhTd4

14:29:05.350253 IP (tos 0x0, ttl 64, id 1535, offset 0, flags [none], proto UDP (17), length 1093)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1065
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---99b4571e317b5877;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKcdJtSCS7uPLcJfKW1ROOB79WBv9G5iEa;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
To: <sip:105@213.87.91.67>;tag=as63421bc7
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19742 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 418

v=0
o=root 2117399007 2117399007 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 14962 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:0lKXAKiA8dtcR20k4u2TCLFhftqcDvad1spIhTd4

14:29:05.749360 IP (tos 0x0, ttl 64, id 1536, offset 0, flags [none], proto UDP (17), length 1091)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1063
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---377b116f1c2ead55;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKg7Mqxu7efdyTC7x0b23uSRrzHUuiPvzR;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=TbymCB77EpJJvrqtLlz8
To: <sip:105@213.87.91.67>;tag=as6f39e0b7
Call-ID: 0a591117-cd17-2c1b-09a9-f87fefe976fa
CSeq: 60448 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 416

v=0
o=root 653978788 653978788 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 18604 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:72SXW3MEuhZt7opeu8/YBFG3LBxMRuxQWhYDAh5O

14:29:07.350220 IP (tos 0x0, ttl 64, id 1537, offset 0, flags [none], proto UDP (17), length 1093)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1065
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---99b4571e317b5877;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKcdJtSCS7uPLcJfKW1ROOB79WBv9G5iEa;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
To: <sip:105@213.87.91.67>;tag=as63421bc7
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19742 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 418

v=0
o=root 2117399007 2117399007 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 14962 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:0lKXAKiA8dtcR20k4u2TCLFhftqcDvad1spIhTd4

14:29:09.750184 IP (tos 0x0, ttl 64, id 1538, offset 0, flags [none], proto UDP (17), length 1091)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1063
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---377b116f1c2ead55;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKg7Mqxu7efdyTC7x0b23uSRrzHUuiPvzR;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=TbymCB77EpJJvrqtLlz8
To: <sip:105@213.87.91.67>;tag=as6f39e0b7
Call-ID: 0a591117-cd17-2c1b-09a9-f87fefe976fa
CSeq: 60448 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 416

v=0
o=root 653978788 653978788 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 18604 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:72SXW3MEuhZt7opeu8/YBFG3LBxMRuxQWhYDAh5O

14:29:10.589683 IP (tos 0x68, ttl 63, id 9819, offset 0, flags [none], proto UDP (17), length 372)
192.168.0.128.sip > asterisk.corp.local.sip: SIP, length: 344
BYE sip:8000@192.168.0.9:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.128:5060;branch=z9hG4bK-b673983a
From: <sip:105@192.168.0.128>;tag=def855099b733efai0
To: <sip:8000@192.168.0.9>;tag=as459babd6
Call-ID: 5ae31da3489d562a5bc81b4c2f507b3d@192.168.0.9:5060
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Cisco/SPA504G-7.5.4
Content-Length: 0

14:29:10.589956 IP (tos 0x0, ttl 64, id 56947, offset 0, flags [none], proto UDP (17), length 461)
asterisk.corp.local.sip > 192.168.0.128.sip: SIP, length: 433
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.128:5060;branch=z9hG4bK-b673983a;received=192.168.0.128
From: <sip:105@192.168.0.128>;tag=def855099b733efai0
To: <sip:8000@192.168.0.9>;tag=as459babd6
Call-ID: 5ae31da3489d562a5bc81b4c2f507b3d@192.168.0.9:5060
CSeq: 101 BYE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

14:29:11.349788 IP (tos 0x0, ttl 64, id 1539, offset 0, flags [none], proto UDP (17), length 1093)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1065
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---99b4571e317b5877;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKcdJtSCS7uPLcJfKW1ROOB79WBv9G5iEa;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
To: <sip:105@213.87.91.67>;tag=as63421bc7
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19742 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 418

v=0
o=root 2117399007 2117399007 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 14962 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:0lKXAKiA8dtcR20k4u2TCLFhftqcDvad1spIhTd4

14:29:13.749699 IP (tos 0x0, ttl 64, id 1540, offset 0, flags [none], proto UDP (17), length 1091)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1063
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---377b116f1c2ead55;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKg7Mqxu7efdyTC7x0b23uSRrzHUuiPvzR;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=TbymCB77EpJJvrqtLlz8
To: <sip:105@213.87.91.67>;tag=as6f39e0b7
Call-ID: 0a591117-cd17-2c1b-09a9-f87fefe976fa
CSeq: 60448 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 416

v=0
o=root 653978788 653978788 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 18604 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:72SXW3MEuhZt7opeu8/YBFG3LBxMRuxQWhYDAh5O

14:29:15.349517 IP (tos 0x0, ttl 64, id 1541, offset 0, flags [none], proto UDP (17), length 1093)
asterisk.corp.local.sip > my.sip-provider.com.8060: SIP, length: 1065
SIP/2.0 200 OK
Via: SIP/2.0/UDP 86.152.30.40:8060;branch=z9hG4bK-524287-1---99b4571e317b5877;received=86.152.30.40;rport=8060
Via: SIP/2.0/TCP 213.87.91.67:1049;branch=z9hG4bKcdJtSCS7uPLcJfKW1ROOB79WBv9G5iEa;rport=1049;received=213.87.91.67
From: <sip:8000@213.87.91.67>;tag=Dio3q6rcpeJVNFDIafcS
To: <sip:105@213.87.91.67>;tag=as63421bc7
Call-ID: 3fc3695c-0efe-75e1-c877-a7376e0fca6c
CSeq: 19742 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:105@192.168.0.9:5060>
Content-Type: application/sdp
Content-Length: 418

v=0
o=root 2117399007 2117399007 IN IP4 192.168.0.9
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.9
t=0 0
m=audio 14962 RTP/SAVPF 0 0 8 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:0lKXAKiA8dtcR20k4u2TCLFhftqcDvad1spIhTd4

awsswa · Сообщение **awsswa** » 29 янв 2013, 18:45

Так, а asterisk собран с правильно собран ?
http://www.slideshare.net/sanjayws/webrtc-asterisk-11

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