Почему Webrtc Asterisk нет звука при входящих?

Question

[Enj0y]

Стоит Asterisk 16.5.1, Freepbx 15.0
Пробовал разные библиотеки, исходящая связь всё ок, входящая нет звука на WebRTC клиенте, всё остальное есть.
DTLS, rtcp Mux, включёны, werbrtc работает на ws и проксируется nginx wss => ws

Comments

[Valentin Barbolin]

Поставь sngrep и скидывай дамп звонка.

[Enj0y]

Andrey Barbolin,

2019/09/08 23:19:49.649826 127.0.0.1:8088 -> 127.0.0.1:59386
INVITE sip:[e-mail];transport=ws SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK26864f4f;rport
Max-Forwards: 70
From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
To: <sip:[e-mail];transport=ws>
Contact: <sip:101@127.0.0.1:5166;transport=ws>
Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
CSeq: 102 INVITE
User-Agent: server
Date: Sun, 08 Sep 2019 20:19:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Call-Info: <sip:ip>;answer-after=0
P-Asserted-Identity: "79999999999" <sip:101@127.0.0.1>
Content-Type: application/sdp
Content-Length: 697

v=0
o=root 356291339 356291339 IN IP4 127.0.0.1
s=Asterisk PBX 16.5.1
c=IN IP4 127.0.0.1
t=0 0
m=audio 14436 RTP/SAVPF 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=ice-ufrag:52c9d2671572a2395d96fed851dd58f2
a=ice-pwd:62c9737c1f1c1af96da6020d19dd8075
a=candidate:Hbca5df9e 1 UDP 2130706431 188.165.***.*** 14436 typ host
a=candidate:Hbca5df9e 2 UDP 2130706430 188.165.***.*** 14437 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 40:62:F0:F5:32:14:09:77:39:9C:08:66:24:E7:66:24:8D:DD:62:79:01:89:52:85:08:78:91:B9:36:9C:3C:39
a=rtcp-mux
a=sendrecv


2019/09/08 23:19:49.712541 127.0.0.1:59386 -> 127.0.0.1:8088
SIP/2.0 100 Trying
Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK26864f4f;rport
To: <sip:[e-mail];transport=ws>
From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
CSeq: 102 INVITE
Supported: outbound
User-Agent: SIP.js/0.7.8
Content-Length: 0



2019/09/08 23:19:49.727988 127.0.0.1:59386 -> 127.0.0.1:8088
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK26864f4f;rport
To: <sip:[e-mail];transport=ws>;tag=omq79k93vm
From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
CSeq: 102 INVITE
Contact: <sip:[e-mail];transport=ws>
Supported: outbound
User-Agent: SIP.js/0.7.8
Content-Length: 0



2019/09/08 23:19:49.756947 127.0.0.1:8088 -> 127.0.0.1:59386
ACK sip:[e-mail];transport=ws SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK1f58ea11;rport
Max-Forwards: 70
From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
To: <sip:[e-mail];transport=ws>;tag=omq79k93vm
Contact: <sip:101@127.0.0.1:5166;transport=ws>
Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
CSeq: 102 ACK
User-Agent: server
Content-Length: 0



2019/09/08 23:19:49.857263 127.0.0.1:8088 -> 127.0.0.1:59386
INVITE sip:[e-mail];transport=ws SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK214a3abf;rport
Max-Forwards: 70
From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
To: <sip:[e-mail];transport=ws>;tag=omq79k93vm
Contact: <sip:101@127.0.0.1:5166;transport=ws>
Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
CSeq: 103 INVITE
User-Agent: server
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "Александр" <sip:89999999999@127.0.0.1>
Content-Type: application/sdp
Content-Length: 697

v=0
o=root 356291339 356291340 IN IP4 127.0.0.1
s=Asterisk PBX 16.5.1
c=IN IP4 127.0.0.1
t=0 0
m=audio 14436 RTP/SAVPF 0 8 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=ice-ufrag:52c9d2671572a2395d96fed851dd58f2
a=ice-pwd:62c9737c1f1c1af96da6020d19dd8075
a=candidate:Hbca5df9e 1 UDP 2130706431 188.165.***.*** 14436 typ host
a=candidate:Hbca5df9e 2 UDP 2130706430 188.165.***.*** 14437 typ host
a=connection:new
a=setup:passive
a=fingerprint:SHA-256 40:62:F0:F5:32:14:09:77:39:9C:08:66:24:E7:66:24:8D:DD:62:79:01:89:52:85:08:78:91:B9:36:9C:3C:39
a=rtcp-mux
a=sendrecv


2019/09/08 23:19:49.930674 127.0.0.1:59386 -> 127.0.0.1:8088
SIP/2.0 100 Trying
Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK214a3abf;rport
To: <sip:[e-mail];transport=ws>;tag=omq79k93vm
From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
CSeq: 103 INVITE
Supported: outbound
User-Agent: SIP.js/0.7.8
Content-Length: 0



2019/09/08 23:19:49.940433 127.0.0.1:8088 -> 127.0.0.1:59386
ACK sip:[e-mail];transport=ws SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK67655c18;rport
Max-Forwards: 70
From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
To: <sip:[e-mail];transport=ws>;tag=omq79k93vm
Contact: <sip:101@127.0.0.1:5166;transport=ws>
Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
CSeq: 103 ACK
User-Agent: server
Content-Length: 0



2019/09/08 23:19:58.139024 127.0.0.1:8088 -> 127.0.0.1:59386
BYE sip:[e-mail];transport=ws SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK37e054da;rport
Max-Forwards: 70
From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
To: <sip:[e-mail];transport=ws>;tag=omq79k93vm
Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
CSeq: 104 BYE
User-Agent: server
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0



2019/09/08 23:19:58.203982 127.0.0.1:59386 -> 127.0.0.1:8088
SIP/2.0 200 OK
Via: SIP/2.0/WS 127.0.0.1:5166;branch=z9hG4bK37e054da;rport
To: <sip:[e-mail];transport=ws>;tag=omq79k93vm
From: "79999999999" <sip:101@127.0.0.1:5166>;tag=as5bdd7a38
Call-ID: 45c330626f23227e638200ef0fefc8c6@127.0.0.1:5166
CSeq: 104 BYE
Supported: outbound
User-Agent: SIP.js/0.7.8
Content-Length: 0

[Enj0y]

Дамп INVITE пару секунд, там сразу идёт приветствие, так что звук должен быть. К слову, ни трубки ни софт фоны с включенным DTLS входящие вообще не берут, возможно не поддерживают, либо дело в DTLS.

[Enj0y]

Попробовал напрямую к WS отключив в браузере блок. всё равно нет звука, я даже не понимаю куда копать...

Answer 1

[Enj0y]

Может дело в проксировании nginx? я пытался настроить сертификат на WSS asterisk, всегда ошибки, плюнул и оставил WS с проксированием Nginx

Comments

[Enj0y]

Выяснил что только через originate нет звука