Зарегистрирован: 07 фев 2017 18:05
Сообщения: 5
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Здравствуйте, требуется помощь. Звонок с WebRtc клиента обрывается после снятия трубки. Отвечающий абонент может быть как местным для FreeSwitch, так и на другом сервере. Такое же поведение при звонке на автоотвечик FreeSwitch. При этом, браузерный клиент получает BY от FreeSwitch с причиной DESTINATION_OUT_OF_ORDER, и в логе FreeSwitch видно ошибку DTLS: [ERR] switch_rtp.c:2917 audio Handshake failure 1Обратный звонок на WebRtc абонента проходит успешно. Проверка openssl s_client на сервере проходит без ошибок. Ситуация возникла еще в конце 2016 года, но ее удавалось лечить мелкими конфигурационными ухищрениями. С конца января после обновления версии Firefox ситуация выглядит стабильно неизлечимой. Исторически было два проекта: один с честным доменным именем сервера( где регистрировались абоненты и от куда они звонили), второй с локальным IP адресом. В 2016 WAN-овский проект жил с честным сертификатом и работал и с Chrome и с Firefox. Lan-овский проект упорно работал только с Firefox, а в Chrome была ситуация аналогичная указанной в теме обсуждения. В 2017 закончился срок использования сертификата в WAN-проекте и после установки самоподписного, его работа полностью восстановилась. На обоих серверах стоит одинаковое ПО(отличие только в нумерации в дайалпланах): Linux deb7-32bit 3.2.0-4-686-pae #1 SMP Debian 3.2.81-1 i686 OpenSSL 1.0.1t 3 May 2016 (Library: OpenSSL 1.0.1e 11 Feb 2013) FreeSWITCH Version 1.4.26-37~32bit (-37 32bit) Клиент выполнен на базе JsSIP.2.0.2, браузер Firefox 51.0.1 (32-бит). JsSIP:UA call() +1m jssip-2.0.2m.js:22556:8
JsSIP:RTCSession new +1ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession connect() +1ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession newRTCSession +9ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession session connecting +181ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession createLocalDescription() +1ms jssip-2.0.2m.js:22556:8
JsSIP:Transport send() +45ms jssip-2.0.2m.js:22556:8
JsSIP:Transport sending message:
INVITE sip:20007@192.168.20.52 SIP/2.0
Via: SIP/2.0/WSS purf14spvc3p.invalid;branch=z9hG4bK4479673
Max-Forwards: 69
To: <sip:20007@192.168.20.52>
From: "720002" <sip:720002@192.168.20.52>;tag=6te7a5uili
Call-ID: i23900k709tkpf41hvhi
CSeq: 5839 INVITE
X-Can-Renegotiate: false
Contact: <sip:dheq7pv9@purf14spvc3p.invalid;transport=ws;ob>
Content-Type: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: ice,replaces,outbound
User-Agent: JsSIP 2.0.2
Content-Length: 1297
v=0
o=mozilla...THIS_IS_SDPARTA-51.0.1 7695237729787045709 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 E9:22:52:20:FE:24:B6:D8:03:05:38:08:0D:E8:7A:74:7C:B4:71:03:A6:DB:2E:C6:81:9C:80:7C:C8:DE:00:E2
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 50327 UDP/TLS/RTP/SAVPF 109 9 0 8
c=IN IP4 192.168.121.1
a=candidate:0 1 UDP 2122121471 192.168.20.87 50326 typ host
a=candidate:1 1 UDP 2122252543 192.168.121.1 50327 typ host
a=candidate:2 1 UDP 2122187007 192.168.137.1 50328 typ host
a=candidate:0 2 UDP 2122121470 192.168.20.87 50329 typ host
a=candidate:1 2 UDP 2122252542 192.168.121.1 50330 typ host
a=candidate:2 2 UDP 2122187006 192.168.137.1 50331 typ host
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=ice-pwd:67a97817cdd327e585ebea100c9452a6
a=ice-ufrag:b1523304
a=mid:sdparta_0
a=msid:{6f58105f-d727-47ce-8a21-da2bdbf410b6} {ae6083b9-f6af-4fd7-bd3e-ef63e837f1b1}
a=rtcp:50330 IN IP4 192.168.121.1
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=setup:actpass
a=ssrc:3064931087 cname:{7fb32195-4a58-4353-9bc1-1ca5b69ec9ad}
+2ms jssip-2.0.2m.js:22556:8
JsSIP:WebSocketInterface send() +0ms jssip-2.0.2m.js:22556:8
JsSIP:WebSocketInterface received WebSocket message +7ms jssip-2.0.2m.js:22556:8
JsSIP:Transport received text message:
SIP/2.0 100 Trying
Via: SIP/2.0/WSS purf14spvc3p.invalid;branch=z9hG4bK4479673;received=127.0.0.1;rport=46480
From: "720002" <sip:720002@192.168.20.52>;tag=6te7a5uili
To: <sip:20007@192.168.20.52>
Call-ID: i23900k709tkpf41hvhi
CSeq: 5839 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.4.26-37~32bit
Content-Length: 0
+1ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession receiveInviteResponse() +6ms jssip-2.0.2m.js:22556:8
JsSIP:WebSocketInterface received WebSocket message +268ms jssip-2.0.2m.js:22556:8
JsSIP:Transport received text message:
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS purf14spvc3p.invalid;branch=z9hG4bK4479673;received=127.0.0.1;rport=46480
From: "720002" <sip:720002@192.168.20.52>;tag=6te7a5uili
To: <sip:20007@192.168.20.52>;tag=cS0r5FrUv2vSB
Call-ID: i23900k709tkpf41hvhi
CSeq: 5839 INVITE
Contact: <sip:20007@192.168.20.52:5062;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.26-37~32bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0
Remote-Party-ID: "Outbound Call" <sip:20007@192.168.20.52>;party=calling;privacy=off;screen=no
+1ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession receiveInviteResponse() +11ms jssip-2.0.2m.js:22556:8
JsSIP:Dialog new UAC dialog created with status EARLY +1ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession session progress +0ms jssip-2.0.2m.js:22556:8
JsSIP:WebSocketInterface received WebSocket message +737ms jssip-2.0.2m.js:22556:8
JsSIP:Transport received text message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS purf14spvc3p.invalid;branch=z9hG4bK4479673;received=127.0.0.1;rport=46480
From: "720002" <sip:720002@192.168.20.52>;tag=6te7a5uili
To: <sip:20007@192.168.20.52>;tag=cS0r5FrUv2vSB
Call-ID: i23900k709tkpf41hvhi
CSeq: 5839 INVITE
Contact: <sip:20007@192.168.20.52:5062;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.4.26-37~32bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 843
Remote-Party-ID: "Outbound Call" <sip:20007@192.168.20.52>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1486517468 1486517469 IN IP4 192.168.20.52
s=FreeSWITCH
c=IN IP4 192.168.20.52
t=0 0
a=msid-semantic: WMS fyQbFCUtLfv3F8MFnfArQMm0F9DByvLE
m=audio 21202 UDP/TLS/RTP/SAVPF 109
a=rtpmap:109 opus/48000/2
a=fmtp:109 useinbandfec=1; maxplaybackrate=48000; stereo=1
a=ptime:20
a=fingerprint:sha-256 D6:E6:DD:D9:EE:EB:4A:B7:D8:4E:48:F6:8A:82:EC:8F:EB:F2:B0:E8:3B:C9:33:3F:FC:6B:89:73:97:8A:88:3A
a=setup:active
a=rtcp-mux
a=rtcp:21202 IN IP4 192.168.20.52
a=ssrc:250089229 cname:UrFpTqIvv4o120xg
a=ssrc:250089229 msid:fyQbFCUtLfv3F8MFnfArQMm0F9DByvLE a0
a=ssrc:250089229 mslabel:fyQbFCUtLfv3F8MFnfArQMm0F9DByvLE
a=ssrc:250089229 label:fyQbFCUtLfv3F8MFnfArQMm0F9DByvLEa0
a=ice-ufrag:5XxMAJD5J6x8r8Cb
a=ice-pwd:p8r9oZuKTJCMzZTvNh7eRA9q
a=candidate:5989554918 1 udp 659136 192.168.20.52 21202 typ host generation 0
+1ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession receiveInviteResponse() +10ms jssip-2.0.2m.js:22556:8
JsSIP:Dialog dialog i23900k709tkpf41hvhi6te7a5uilicS0r5FrUv2vSB changed to CONFIRMED state +1ms jssip-2.0.2m.js:22556:8
Tryit: addstream() gui.js:146:9
addstream 0 -i23900k709tkpf41hvhi6te7a5uili gui.js:153:19
JsSIP:RTCSession session accepted +25ms jssip-2.0.2m.js:22556:8
accepted stream 0 -i23900k709tkpf41hvhi6te7a5uili gui.js:121:21
JsSIP:RTCSession sendRequest() +5ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession:Request new | ACK +1ms jssip-2.0.2m.js:22556:8
JsSIP:Transport send() +3ms jssip-2.0.2m.js:22556:8
JsSIP:Transport sending message:
ACK sip:20007@192.168.20.52:5062;transport=udp SIP/2.0
Via: SIP/2.0/WSS purf14spvc3p.invalid;branch=z9hG4bK1447185
Max-Forwards: 69
To: <sip:20007@192.168.20.52>;tag=cS0r5FrUv2vSB
From: "720002" <sip:720002@192.168.20.52>;tag=6te7a5uili
Call-ID: i23900k709tkpf41hvhi
CSeq: 5839 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 2.0.2
Content-Length: 0
+0ms jssip-2.0.2m.js:22556:8
JsSIP:WebSocketInterface send() +1ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession session confirmed +1ms jssip-2.0.2m.js:22556:8
JsSIP:WebSocketInterface received WebSocket message +104ms jssip-2.0.2m.js:22556:8
JsSIP:Transport received text message:
BYE sip:dheq7pv9@purf14spvc3p.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WSS 127.0.0.1:7443;branch=z9hG4bK4K7QcQtD3rHSS
Max-Forwards: 70
From: <sip:20007@192.168.20.52>;tag=cS0r5FrUv2vSB
To: "720002" <sip:720002@192.168.20.52>;tag=6te7a5uili
Call-ID: i23900k709tkpf41hvhi
CSeq: 102933271 BYE
User-Agent: FreeSWITCH-mod_sofia/1.4.26-37~32bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Reason: Q.850;cause=27;text="DESTINATION_OUT_OF_ORDER"
Content-Length: 0
+1ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession receiveRequest() +6ms jssip-2.0.2m.js:22556:8
JsSIP:Transport send() +1ms jssip-2.0.2m.js:22556:8
JsSIP:Transport sending message:
SIP/2.0 200 OK
Via: SIP/2.0/WSS 127.0.0.1:7443;branch=z9hG4bK4K7QcQtD3rHSS
To: "720002" <sip:720002@192.168.20.52>;tag=6te7a5uili
From: <sip:20007@192.168.20.52>;tag=cS0r5FrUv2vSB
Call-ID: i23900k709tkpf41hvhi
CSeq: 102933271 BYE
Supported: outbound
Content-Length: 0
+0ms jssip-2.0.2m.js:22556:8
JsSIP:WebSocketInterface send() +1ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession session ended +1ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession close() +0ms jssip-2.0.2m.js:22556:8
JsSIP:RTCSession close() | closing local MediaStream +1ms jssip-2.0.2m.js:22556:8
JsSIP:Dialog dialog i23900k709tkpf41hvhi6te7a5uilicS0r5FrUv2vSB deleted +1ms jssip-2.0.2m.js:22556:8 Серверный софт - не мой конек. Буду благодарен любым советам.
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