“WebRTC Error” on incoming jssip call with video
Sergey Tamkovich
Hello everybody,
I’m not sure whether my question is more related to jssip or to webrtc, but i’m desperate so i’ll try here.
I have a door bell, which can initiate sip video calls with ulaw/h264. I’ve installed asterisk and configured it to accept sip endpoints through udp and websocket. Using examples from the internet i’ve built react-native app for android to receive calls using react-native-webrtc and react-native-jssip. It even works if i place an audio call from asterisk to my RN app. I can answer it, audio travels both directions - everything is perfect.
However, if i enable h264 codec on asterisk for RN endpoint - i got an error from webrtc, when i'm trying to answer the call. I googled and tried everything before writing this question, maybe someone can guide me - most likely i’m doing something wrong or missing some important bit.
I would really appreciate any advice,
thank you in advance!
UA init:
let socket = new JsSIP.WebSocketInterface('wss://domofon.fqdn:8089/ws');
let configuration = {
sockets: [socket],
uri: 'sip:webrtc...@domofon.fqdn',
password: 'webrtc_client',
session_timers: false,
};
let ua = new JsSIP.UA(configuration);
Call answer:
const options = {
eventHandlers: eventHandlers,
mediaConstraints: {audio: true, video: true},
pcConfig: {
rtcpMuxPolicy: 'require',
iceServers: [],
},
rtcOfferConstraints: {
offerToReceiveAudio: 1,
offerToReceiveVideo: 1,
},
};
call.answer(options);
Debug output:
INVITE sip:dd6e...@home.ip:40581;transport=ws SIP/2.0
Via: SIP/2.0/WSS asterisk.ip:8089;rport;branch=z9hG4bKPj2f2a463d-2719-44f7-ad7a-4e01e2fbd221;alias
From: <sip:014@domofon-test>;tag=a889e457-1ab9-4f5b-9e7b-4b65977a1062
To: <sip:dd6e...@home.ip>
Contact: <sip:asterisk@domofon-test:5060;transport=ws>
Call-ID: e9402b69-bb8c-4367-adea-7f88a42df219
CSeq: 24000 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.4.0
Content-Type: application/sdp
Content-Length: 1549
v=0
o=- 247016796 247016796 IN IP4 asterisk.ip
s=Asterisk
c=IN IP4 asterisk.ip
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0 video-1
m=audio 16692 UDP/TLS/RTP/SAVPF 0 8 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 52:93:08:22:BD:B0:26:92:86:40:04:A1:BC:79:9F:4C:40:02:D0:EB:85:E8:B8:B3:DF:14:CD:C4:84:BA:3A:CB
a=ice-ufrag:45a51fda68f5269042081a4a3f128834
a=ice-pwd:7ec90f4b39d7070253c507b26cf09df0
a=candidate:Hb00c610e 1 UDP 2130706431 asterisk.ip 16692 typ host
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=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:2114533563 cname:2ec6da81-7aec-4dea-b6a4-1e310b5c3feb
a=msid:52024678-70c5-4442-adbd-7140de561366 361ae9ca-8b75-4822-98aa-b0ec635a11b6
a=rtcp-fb:* transport-cc
a=mid:audio-0
m=video 16692 UDP/TLS/RTP/SAVPF 99
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 52:93:08:22:BD:B0:26:92:86:40:04:A1:BC:79:9F:4C:40:02:D0:EB:85:E8:B8:B3:DF:14:CD:C4:84:BA:3A:CB
a=ice-ufrag:45a51fda68f5269042081a4a3f128834
a=ice-pwd:7ec90f4b39d7070253c507b26cf09df0
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;profile-level-id=428016
a=sendrecv
a=rtcp-mux
a=ssrc:370757493 cname:dbc942ca-ccb3-4b9c-afb6-9b7b47d578fc
a=msid:96f48882-751e-445b-a50d-f0ffbb46d66f 5b4355a0-f43d-4398-91e2-0dd7be184de2
a=rtcp-fb:* transport-cc
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=rtcp-fb:* nack
a=extmap:1 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=mid:video-1
+10s
DEBUG JsSIP:Transport send() +44ms
DEBUG JsSIP:Transport sending message:
SIP/2.0 100 Trying
Via: SIP/2.0/WSS asterisk.ip:8089;rport;branch=z9hG4bKPj2f2a463d-2719-44f7-ad7a-4e01e2fbd221;alias
To: <sip:dd6e...@home.ip>
From: <sip:014@domofon-test>;tag=a889e457-1ab9-4f5b-9e7b-4b65977a1062
Call-ID: e9402b69-bb8c-4367-adea-7f88a42df219
CSeq: 24000 INVITE
Supported: ice,replaces,outbound
Content-Length: 0
+2ms
DEBUG JsSIP:WebSocketInterface send() +47ms
DEBUG JsSIP:RTCSession new +0ms
DEBUG JsSIP:RTCSession init_incoming() +3ms
DEBUG JsSIP:Dialog new UAS dialog created with status EARLY +0ms
DEBUG JsSIP:RTCSession newRTCSession() +9ms
LOG event newRTCSession fired
LOG incoming call
DEBUG JsSIP:WebSocketInterface new() [url:"wss://domofon.fqdn:8089/ws"] +35ms
DEBUG JsSIP:UA new() [configuration:'{"password": "webrtc_client", "session_timers": false, "sockets": [{"_sip_uri": "sip:domofon.fqdn:8089;transport=ws", "_url": "wss://domofon.fqdn:8089/ws", "_via_transport": "WSS", "_ws": null}], "uri": "sip:webrtc...@domofon.fqdn"}'] +11s
DEBUG JsSIP:Transport new() +72ms
WARN JsSIP:ERROR:Transport cannot use TextDecoder: ReferenceError: Can't find variable: TextDecoder +11s
RCTView
View
RCTView
View
d2(RootComponent)@http://localhost:8081/index.bundle?platform=android&dev=true&minify=false&app=com.d2&modulesOnly=false&runModule=true:81851:28
DEBUG JsSIP:UA configuration parameters after validation: +65ms
DEBUG JsSIP:UA - authorization_user: "webrtc_client" +1ms
DEBUG JsSIP:UA - password: NOT SHOWN +0ms
DEBUG JsSIP:UA - realm: null +1ms
DEBUG JsSIP:UA - ha1: NOT SHOWN +0ms
DEBUG JsSIP:UA - authorization_jwt: NOT SHOWN +0ms
DEBUG JsSIP:UA - display_name: null +0ms
DEBUG JsSIP:UA - uri: sip:webrtc...@domofon.fqdn +1ms
DEBUG JsSIP:UA - contact_uri: {"_parameters":{"transport":"ws"},"_headers":{},"_scheme":"sip","_user":"lbrb8gp1","_host":"ogrg28hbjfng.invalid","_port":null} +0ms
DEBUG JsSIP:UA - instance_id: "8762e436-330b-4297-9ce9-d2599d754e50" +0ms
DEBUG JsSIP:UA - use_preloaded_route: false +0ms
DEBUG JsSIP:UA - session_timers: false +1ms
DEBUG JsSIP:UA - session_timers_refresh_method: "UPDATE" +0ms
DEBUG JsSIP:UA - session_timers_force_refresher: false +0ms
DEBUG JsSIP:UA - no_answer_timeout: 60000 +0ms
DEBUG JsSIP:UA - register: true +0ms
DEBUG JsSIP:UA - register_expires: 600 +1ms
DEBUG JsSIP:UA - registrar_server: sip:domofon.fqdn +0ms
DEBUG JsSIP:UA - connection_recovery_max_interval: 30 +0ms
DEBUG JsSIP:UA - connection_recovery_min_interval: 2 +0ms
DEBUG JsSIP:UA - via_host: "ogrg28hbjfng.invalid" +0ms
DEBUG JsSIP:UA start() +1ms
DEBUG JsSIP:Transport connect() +38ms
DEBUG JsSIP:WebSocketInterface connect() +76ms
DEBUG JsSIP:WebSocketInterface connecting to WebSocket wss://domofon.fqdn:8089/ws +0ms
DEBUG JsSIP:WebSocketInterface new() [url:"wss://domofon.fqdn:8089/ws"] +34ms
DEBUG JsSIP:UA new() [configuration:'{"password": "webrtc_client", "session_timers": false, "sockets": [{"_sip_uri": "sip:domofon.fqdn:8089;transport=ws", "_url": "wss://domofon.fqdn:8089/ws", "_via_transport": "WSS", "_ws": null}], "uri": "sip:webrtc...@domofon.fqdn"}'] +40ms
DEBUG JsSIP:Transport new() +46ms
WARN JsSIP:ERROR:Transport cannot use TextDecoder: ReferenceError: Can't find variable: TextDecoder +83ms
RCTView
View
RCTView
View
d2(RootComponent)@http://localhost:8081/index.bundle?platform=android&dev=true&minify=false&app=com.d2&modulesOnly=false&runModule=true:81851:28
DEBUG JsSIP:UA configuration parameters after validation: +18ms
DEBUG JsSIP:UA - authorization_user: "webrtc_client" +1ms
DEBUG JsSIP:UA - password: NOT SHOWN +0ms
DEBUG JsSIP:UA - realm: null +0ms
DEBUG JsSIP:UA - ha1: NOT SHOWN +0ms
DEBUG JsSIP:UA - authorization_jwt: NOT SHOWN +0ms
DEBUG JsSIP:UA - display_name: null +1ms
DEBUG JsSIP:UA - uri: sip:webrtc...@domofon.fqdn +1ms
DEBUG JsSIP:UA - contact_uri: {"_parameters":{"transport":"ws"},"_headers":{},"_scheme":"sip","_user":"oin82sv2","_host":"ogrg28hbjfng.invalid","_port":null} +0ms
DEBUG JsSIP:UA - instance_id: "de73e18d-16d5-44b0-9243-33593c147079" +1ms
DEBUG JsSIP:UA - use_preloaded_route: false +0ms
DEBUG JsSIP:UA - session_timers: false +1ms
DEBUG JsSIP:UA - session_timers_refresh_method: "UPDATE" +0ms
DEBUG JsSIP:UA - session_timers_force_refresher: false +0ms
DEBUG JsSIP:UA - no_answer_timeout: 60000 +0ms
DEBUG JsSIP:UA - register: true +1ms
DEBUG JsSIP:UA - register_expires: 600 +0ms
DEBUG JsSIP:UA - registrar_server: sip:domofon.fqdn +0ms
DEBUG JsSIP:UA - connection_recovery_max_interval: 30 +0ms
DEBUG JsSIP:UA - connection_recovery_min_interval: 2 +2ms
DEBUG JsSIP:UA - via_host: "ogrg28hbjfng.invalid" +1ms
DEBUG JsSIP:UA start() +0ms
DEBUG JsSIP:Transport connect() +20ms
DEBUG JsSIP:WebSocketInterface connect() +31ms
DEBUG JsSIP:WebSocketInterface connecting to WebSocket wss://domofon.fqdn:8089/ws +1ms
DEBUG JsSIP:Transport send() +18ms
DEBUG JsSIP:Transport sending message:
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS asterisk.ip:8089;rport;branch=z9hG4bKPj2f2a463d-2719-44f7-ad7a-4e01e2fbd221;alias
To: <sip:dd6e...@home.ip>;tag=00q1dm9b27
From: <sip:014@domofon-test>;tag=a889e457-1ab9-4f5b-9e7b-4b65977a1062
Call-ID: e9402b69-bb8c-4367-adea-7f88a42df219
CSeq: 24000 INVITE
Contact: <sip:dd6e...@ogrg28hbjfng.invalid;transport=ws>
Supported: ice,replaces,outbound
Content-Length: 0
+1ms
DEBUG JsSIP:WebSocketInterface send() +18ms
DEBUG JsSIP:RTCSession session progress +168ms
DEBUG JsSIP:RTCSession emit "progress" +1ms
DEBUG JsSIP:WebSocketInterface WebSocket wss://domofon.fqdn:8089/ws connected +236ms
LOG Answering call
DEBUG JsSIP:RTCSession answer() +2s
DEBUG JsSIP:Dialog dialog e9402b69-bb8c-4367-adea-7f88a42df21900q1dm9b27a889e457-1ab9-4f5b-9e7b-4b65977a1062 changed to CONFIRMED state +3s
DEBUG JsSIP:RTCSession emit "peerconnection" +59ms
LOG event session.peerconnection fired
DEBUG JsSIP:RTCSession emit "sdp" +87ms
LOG event session.connection.negotiationneeded fired
LOG {"bubbles": false, "cancelable": false, "currentTarget": {"_localStreams": [[MediaStream]], "_peerConnectionId": 0, "_remoteStreams": [], "_subscriptions": [[Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object]], "connectionState": "new", "iceConnectionState": "new", "iceGatheringState": "new", "ontrack": [Function anonymous], "signalingState": "stable"}, "eventPhase": 2, "isTrusted": false, "target": {"_localStreams": [[MediaStream]], "_peerConnectionId": 0, "_remoteStreams": [], "_subscriptions": [[Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object]], "connectionState": "new", "iceConnectionState": "new", "iceGatheringState": "new", "ontrack": [Function anonymous], "signalingState": "stable"}, "timeStamp": 1633188822286, "type": "negotiationneeded"}
LOG event session.connection.signalingstatechange fired
LOG {"bubbles": false, "cancelable": false, "currentTarget": {"_localStreams": [[MediaStream]], "_peerConnectionId": 0, "_remoteStreams": [], "_subscriptions": [[Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object]], "connectionState": "new", "iceConnectionState": "new", "iceGatheringState": "new", "ontrack": [Function anonymous], "signalingState": "have-remote-offer"}, "eventPhase": 2, "isTrusted": false, "target": {"_localStreams": [[MediaStream]], "_peerConnectionId": 0, "_remoteStreams": [], "_subscriptions": [[Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object], [Object]], "connectionState": "new", "iceConnectionState": "new", "iceGatheringState": "new", "ontrack": [Function anonymous], "signalingState": "have-remote-offer"}, "timeStamp": 1633188822309, "type": "signalingstatechange"}
DEBUG JsSIP:Transport send() +2s
DEBUG JsSIP:Transport sending message:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS asterisk.ip:8089;rport;branch=z9hG4bKPj2f2a463d-2719-44f7-ad7a-4e01e2fbd221;alias
To: <sip:dd6e...@home.ip>;tag=00q1dm9b27
From: <sip:014@domofon-test>;tag=a889e457-1ab9-4f5b-9e7b-4b65977a1062
Call-ID: e9402b69-bb8c-4367-adea-7f88a42df219
CSeq: 24000 INVITE
Supported: ice,replaces,outbound
Content-Length: 0
+1ms
DEBUG JsSIP:WebSocketInterface send() +2s
DEBUG JsSIP:RTCSession session failed +47ms
DEBUG JsSIP:RTCSession emit "_failed" +2ms
DEBUG JsSIP:RTCSession close() +1ms
DEBUG JsSIP:RTCSession close() | closing local MediaStream +1ms
DEBUG JsSIP:Dialog dialog e9402b69-bb8c-4367-adea-7f88a42df21900q1dm9b27a889e457-1ab9-4f5b-9e7b-4b65977a1062 deleted +215ms
DEBUG JsSIP:RTCSession emit "failed" +23ms
LOG event session.failed fired
LOG {"cause": "WebRTC Error", "message": null, "originator": "system"}
DEBUG JsSIP:WebSocketInterface new() [url:"wss://domofon.fqdn:8089/ws"] +33ms
DEBUG JsSIP:UA new() [configuration:'{"password": "webrtc_client", "session_timers": false, "sockets": [{"_sip_uri": "sip:domofon.fqdn:8089;transport=ws", "_url": "wss://domofon.fqdn:8089/ws", "_via_transport": "WSS", "_ws": null}], "uri": "sip:webrtc...@domofon.fqdn"}'] +3s
DEBUG JsSIP:Transport new() +43ms
WARN JsSIP:ERROR:Transport cannot use TextDecoder: ReferenceError: Can't find variable: TextDecoder +3s
RCTView
View
RCTView
View
d2(RootComponent)@http://localhost:8081/index.bundle?platform=android&dev=true&minify=false&app=com.d2&modulesOnly=false&runModule=true:81851:28
DEBUG JsSIP:UA configuration parameters after validation: +20ms
DEBUG JsSIP:UA - authorization_user: "webrtc_client" +1ms
DEBUG JsSIP:UA - password: NOT SHOWN +1ms
DEBUG JsSIP:UA - realm: null +0ms
DEBUG JsSIP:UA - ha1: NOT SHOWN +1ms
DEBUG JsSIP:UA - authorization_jwt: NOT SHOWN +0ms
DEBUG JsSIP:UA - display_name: null +2ms
DEBUG JsSIP:UA - uri: sip:webrtc...@domofon.fqdn +0ms
DEBUG JsSIP:UA - contact_uri: {"_parameters":{"transport":"ws"},"_headers":{},"_scheme":"sip","_user":"45vkg9d7","_host":"ogrg28hbjfng.invalid","_port":null} +2ms
DEBUG JsSIP:UA - instance_id: "f0fc2e3c-9113-4b25-8a25-7149694572eb" +1ms
DEBUG JsSIP:UA - use_preloaded_route: false +0ms
DEBUG JsSIP:UA - session_timers: false +0ms
DEBUG JsSIP:UA - session_timers_refresh_method: "UPDATE" +1ms
DEBUG JsSIP:UA - session_timers_force_refresher: false +0ms
DEBUG JsSIP:UA - no_answer_timeout: 60000 +1ms
DEBUG JsSIP:UA - register: true +0ms
DEBUG JsSIP:UA - register_expires: 600 +1ms
DEBUG JsSIP:UA - registrar_server: sip:domofon.fqdn +1ms
DEBUG JsSIP:UA - connection_recovery_max_interval: 30 +1ms
DEBUG JsSIP:UA - connection_recovery_min_interval: 2 +1ms
DEBUG JsSIP:UA - via_host: "ogrg28hbjfng.invalid" +0ms
DEBUG JsSIP:UA start() +0ms
DEBUG JsSIP:Transport connect() +27ms
DEBUG JsSIP:WebSocketInterface connect() +37ms
DEBUG JsSIP:WebSocketInterface connecting to WebSocket wss://domofon.fqdn:8089/ws +0ms
WARN JsSIP:ERROR:RTCSession emit "peerconnection:setremotedescriptionfailed" [error:"Failed to set remote offer sdp: Failed to set remote video description send parameters for m-section with mid='video-1'."] +0ms
DEBUG JsSIP:WebSocketInterface received WebSocket message +81ms
DEBUG JsSIP:Transport received text message:
ACK sip:dd6e...@home.ip:40581;transport=ws SIP/2.0
Via: SIP/2.0/WSS asterisk.ip:8089;rport;branch=z9hG4bKPj2f2a463d-2719-44f7-ad7a-4e01e2fbd221;alias
From: <sip:014@domofon-test>;tag=a889e457-1ab9-4f5b-9e7b-4b65977a1062
To: <sip:dd6e...@home.ip>;tag=00q1dm9b27
Call-ID: e9402b69-bb8c-4367-adea-7f88a42df219
CSeq: 24000 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.4.0
Content-Length: 0
+82ms
LOG event session.connection.iceconnectionstatechange fired
Roman Shpount
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Sergey Tamkovich
Unfortunately i can't open your url, google says "document not found".
Roman Shpount
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