No audio over WebRTC with Asterisk 11.7

Asterisk Asterisk Support
1

I’m having what is probably a simple configuration issue that I’m having some trouble tracking down. I’ve set up a fresh Asterisk install (11.7 RC1) and am trying to connect to it over the Internet using WebRTC (using the steps outlined in the docs: wiki.asterisk.org/wiki/display/ … TC+Support), audio only. I’ve tried using both Sipml5 and JSSIP on the client. I turned SIP and RTP debugging on and here’s what I’m seeing:

SIP Registration, server:

[code]== WebSocket connection from ‘71.234.235.104:41935’ for protocol ‘sip’ accepted using version ‘13’

<— SIP read from WS:71.234.235.104:41935 —>
REGISTER sip:162.243.116.82 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKeET8TlwwasyM4k2TLvKkfp9lN2Z6Lq6h;rport
From: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=mmpTqBd0tOnufwWq6FIa
To: "WebRTCClient"sip:WebRTCClient@162.243.116.82
Contact: "WebRTCClient"sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: daa031ec-3d22-088a-5a62-c3af858e7179
CSeq: 19622 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom
Supported: path

<------------->
— (12 headers 0 lines) —
– Registered SIP ‘WebRTCClient’ at 71.234.235.104:41935

<— Transmitting (NAT) to 71.234.235.104:41935 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKeET8TlwwasyM4k2TLvKkfp9lN2Z6Lq6h;received=71.234.235.104;rport=41935
From: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=mmpTqBd0tOnufwWq6FIa
To: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=as4b80b1fd
Call-ID: daa031ec-3d22-088a-5a62-c3af858e7179
CSeq: 19622 REGISTER
Server: Sipml5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 200
Contact: sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws;expires=200
Date: Fri, 06 Dec 2013 17:27:24 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘daa031ec-3d22-088a-5a62-c3af858e7179’ in 32000 ms (Method: REGISTER)[/code]

SIP registration, client:

[code]SIPML5 API version = 1.3.203 SIPml-api.js?svn=179:1
location=http://sipml5.org/call.htm?svn=203# call.htm?svn=203:147
User-Agent=Mozilla/5.0 (Macintosh; Intel Mac OS X 10_9_0) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/31.0.1650.57 Safari/537.36 SIPml-api.js?svn=179:1
WebSocket supported = yes SIPml-api.js?svn=179:1
Navigator friendly name = chrome SIPml-api.js?svn=179:1
OS friendly name = mac SIPml-api.js?svn=179:1
Have WebRTC = yes SIPml-api.js?svn=179:1
Have GUM = yes SIPml-api.js?svn=179:1
Engine initialized SIPml-api.js?svn=179:1
event.returnValue is deprecated. Please use the standard event.preventDefault() instead.
s_websocket_server_url=ws://162.243.116.82:8088/ws SIPml-api.js?svn=179:1
s_sip_outboundproxy_url=(null) SIPml-api.js?svn=179:1
b_rtcweb_breaker_enabled=no SIPml-api.js?svn=179:1
b_click2call_enabled=no SIPml-api.js?svn=179:1
b_early_ims=yes SIPml-api.js?svn=179:1
b_enable_media_stream_cache=no SIPml-api.js?svn=179:1
o_bandwidth={} SIPml-api.js?svn=179:1
o_video_size={} SIPml-api.js?svn=179:1
SIP stack start: proxy=‘ns313841.ovh.net:11060’, realm=‘sip:162.243.116.82’, impi=‘WebRTCClient’, impu=’"WebRTCClient"sip:WebRTCClient@162.243.116.82’ SIPml-api.js?svn=179:1
Connecting to ‘ws://162.243.116.82:8088/ws’ SIPml-api.js?svn=179:1
==stack event = starting SIPml-api.js?svn=179:1
__tsip_transport_ws_onopen SIPml-api.js?svn=179:1
==stack event = started SIPml-api.js?svn=179:1
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister SIPml-api.js?svn=179:1
SEND: REGISTER sip:162.243.116.82 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKeET8TlwwasyM4k2TLvKkfp9lN2Z6Lq6h;rport
From: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=mmpTqBd0tOnufwWq6FIa
To: "WebRTCClient"sip:WebRTCClient@162.243.116.82
Contact: "WebRTCClient"sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: daa031ec-3d22-088a-5a62-c3af858e7179
CSeq: 19622 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom
Supported: path

SIPml-api.js?svn=179:1
==session event = connecting SIPml-api.js?svn=179:1
==session event = sent_request SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=41935;received=71.234.235.104;branch=z9hG4bKeET8TlwwasyM4k2TLvKkfp9lN2Z6Lq6h
From: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=mmpTqBd0tOnufwWq6FIa
To: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=as4b80b1fd
Contact: sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws;expires=200
Call-ID: daa031ec-3d22-088a-5a62-c3af858e7179
CSeq: 19622 REGISTER
Expires: 200
Content-Length: 0
Server: Sipml5
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 06 Dec 2013 17:27:24 GMT;06

SIPml-api.js?svn=179:1
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js?svn=179:1
==session event = connected

State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister SIPml-api.js?svn=179:1
SEND: REGISTER sip:162.243.116.82 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKmO5ynBn7ETDfkXLV5qk8B56M2KXZMhIb;rport
From: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=mmpTqBd0tOnufwWq6FIa
To: "WebRTCClient"sip:WebRTCClient@162.243.116.82
Contact: "WebRTCClient"sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: daa031ec-3d22-088a-5a62-c3af858e7179
CSeq: 19623 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom

SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=41935;received=71.234.235.104;branch=z9hG4bKmO5ynBn7ETDfkXLV5qk8B56M2KXZMhIb
From: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=mmpTqBd0tOnufwWq6FIa
To: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=as078051c5
Contact: sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws;expires=200
Call-ID: daa031ec-3d22-088a-5a62-c3af858e7179
CSeq: 19623 REGISTER
Expires: 200
Content-Length: 0
Server: Sipml5
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 06 Dec 2013 17:29:05 GMT;06

SIPml-api.js?svn=179:1
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js?svn=179:1
==session event = sent_request [/code]

Then, when I try to call into a conference, I get no audio, and RTP logs show a whole lot of this with no packets received:

Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014258, ts 026720, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014259, ts 026880, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014260, ts 027040, len 000164) -- <SIP/WebRTCClient-00000000> Playing 'confbridge-join.gsm' (language 'en') Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014261, ts 027200, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014262, ts 027360, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014263, ts 027520, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014264, ts 027680, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014265, ts 027840, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014266, ts 028000, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014267, ts 028160, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014268, ts 028320, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014269, ts 028480, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014270, ts 028640, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014271, ts 028800, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014272, ts 028960, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014273, ts 029120, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014274, ts 029280, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014275, ts 029440, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014276, ts 029600, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014277, ts 029760, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014278, ts 029920, len 000164) -- <Bridge/0x7ffdb0002378-input> Playing 'confbridge-join.gsm' (language 'en') Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014279, ts 033144, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014280, ts 033304, len 000164) Sent RTP packet to 71.234.235.104:34327 (type 00, seq 014281, ts 033464, len 000164)

And on the client, once I make the call:

[code]State machine: c0000_Started_2_Outgoing_X_oINVITE SIPml-api.js?svn=179:1
PeerConnectionClass = function RTCPeerConnection() { [native code] } SessionDescriptionClass = function RTCSessionDescription() { [native code] } IceCandidateClass = function RTCIceCandidate() { [native code] } SIPml-api.js?svn=179:1
ICE servers:[{“url”:“stun:stun.l.google.com:19302”},{“url”:“stun:stun.counterpath.net:3478”},{“url”:“stun:numb.viagenie.ca:3478”}] SIPml-api.js?svn=179:1
==stack event = m_permission_requested SIPml-api.js?svn=179:1
==session event = connecting SIPml-api.js?svn=179:1
onGetUserMediaSuccess SIPml-api.js?svn=179:1
createOffer SIPml-api.js?svn=179:1
onCreateSdpSuccess SIPml-api.js?svn=179:1
==stack event = m_permission_accepted SIPml-api.js?svn=179:1
==session event = m_stream_audio_local_added SIPml-api.js?svn=179:1
onSetLocalDescriptionSuccess SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
onIceCandidate = undefined SIPml-api.js?svn=179:1
ICE GATHERING COMPLETED! SIPml-api.js?svn=179:1
onIceGatheringCompleted SIPml-api.js?svn=179:1
SEND: INVITE sip:602@162.243.116.82 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKVlNAwXvOgDUIgdfsf0psNxlw2zMhZbcu;rport
From: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=8Se4MBGpG5kr6JydAmtl
To: sip:602@162.243.116.82
Contact: "WebRTCClient"sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 6d618654-8e4f-cf0d-c7d9-2c70cc73050b
CSeq: 24894 INVITE
Content-Type: application/sdp
Content-Length: 1826
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom

v=0
o=- 8391975185657087000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS dPiG5yvOH9jdjI81oipXlDLGeCodt6LHiMie
m=audio 33153 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 71.234.235.104
a=rtcp:33153 IN IP4 71.234.235.104
a=candidate:1407141326 1 udp 2113937151 10.0.1.2 55659 typ host generation 0
a=candidate:1407141326 2 udp 2113937151 10.0.1.2 55659 typ host generation 0
a=candidate:1824104935 1 udp 1845501695 71.234.235.104 33153 typ srflx raddr 10.0.1.2 rport 55659 generation 0
a=candidate:1824104935 2 udp 1845501695 71.234.235.104 33153 typ srflx raddr 10.0.1.2 rport 55659 generation 0
a=candidate:492615998 1 tcp 1509957375 10.0.1.2 0 typ host generation 0
a=candidate:492615998 2 tcp 1509957375 10.0.1.2 0 typ host generation 0
a=ice-ufrag:HviNji6VV5zVVhrs
a=ice-pwd:A+5Kpe9H3sy5Or27nq80BhPb
a=ice-options:google-ice
a=fingerprint:sha-256 9C:ED:BE:E4:29:C0:56:74:69:31:58:2F:C7:1D:BC:9D:08:86:DA:06:AE:E6:F1:AE:6E:34:C7:23:04:AE:99:AD
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:mPQn5bRO7dsBZyEWOTJqsCPVT1fxvwgke69zrB1O
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:cT49WUHc3TopGvnQbhuG/hSujFTwAiCYTvoOZuY5
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:988313808 cname:6IbvpvGRByBcKkIe
a=ssrc:988313808 msid:dPiG5yvOH9jdjI81oipXlDLGeCodt6LHiMie dPiG5yvOH9jdjI81oipXlDLGeCodt6LHiMiea0
a=ssrc:988313808 mslabel:dPiG5yvOH9jdjI81oipXlDLGeCodt6LHiMie
a=ssrc:988313808 label:dPiG5yvOH9jdjI81oipXlDLGeCodt6LHiMiea0
SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=48954;received=71.234.235.104;branch=z9hG4bKVlNAwXvOgDUIgdfsf0psNxlw2zMhZbcu
From: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=8Se4MBGpG5kr6JydAmtl
To: sip:602@162.243.116.82
Contact: sip:602@162.243.116.82:5060;transport=WS
Call-ID: 6d618654-8e4f-cf0d-c7d9-2c70cc73050b
CSeq: 24894 INVITE
Content-Length: 0
Server: Sipml5
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

SIPml-api.js?svn=179:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js?svn=179:1
==session event = i_ao_request SIPml-api.js?svn=179:1
__tsip_transport_ws_onmessage SIPml-api.js?svn=179:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=48954;received=71.234.235.104;branch=z9hG4bKVlNAwXvOgDUIgdfsf0psNxlw2zMhZbcu
From: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=8Se4MBGpG5kr6JydAmtl
To: sip:602@162.243.116.82;tag=as0ca86c53
Contact: sip:602@162.243.116.82:5060;transport=WS
Call-ID: 6d618654-8e4f-cf0d-c7d9-2c70cc73050b
CSeq: 24894 INVITE
Content-Type: application/sdp
Content-Length: 376
Server: Sipml5
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer

v=0
o=root 96033238 96033238 IN IP4 162.243.116.82
s=Asterisk PBX 11.7.0-rc1
c=IN IP4 162.243.116.82
t=0 0
m=audio 19194 RTP/SAVPF 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:XOOBAO0T+B+69kwKTM1xP6Wfd/HfnPFzmDV9o2vZ
SIPml-api.js?svn=179:1
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE SIPml-api.js?svn=179:1
setRemoteDescription(answer)
v=0
o=root 96033238 96033238 IN IP4 162.243.116.82
s=Asterisk PBX 11.7.0-rc1
c=IN IP4 162.243.116.82
t=0 0
m=audio 19194 RTP/SAVPF 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:XOOBAO0T+B+69kwKTM1xP6Wfd/HfnPFzmDV9o2vZ
SIPml-api.js?svn=179:1
SEND: ACK sip:602@162.243.116.82:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKyNKVMdWe1AhBWtnNon4A;rport
From: "WebRTCClient"sip:WebRTCClient@162.243.116.82;tag=8Se4MBGpG5kr6JydAmtl
To: sip:602@162.243.116.82;tag=as0ca86c53
Contact: "WebRTCClient"sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 6d618654-8e4f-cf0d-c7d9-2c70cc73050b
CSeq: 24894 ACK
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom

SIPml-api.js?svn=179:1
__on_add_stream SIPml-api.js?svn=179:1
onSetRemoteDescriptionSuccess SIPml-api.js?svn=179:1
==session event = m_early_media SIPml-api.js?svn=179:1
==session event = connected SIPml-api.js?svn=179:1
==session event = m_stream_audio_remote_added [/code]

Here’s my sip.conf:

[general]
context = public
realm = 162.243.116.82
bindaddr = 0.0.0.0
useragent = Sipml5

[webrtc-template](!)
type = friend
context = webrtc
host = dynamic
transport = ws,wss
avpf = yes
encryption = yes

[WebRTCClient](webrtc-template)
callerid = "WebRTCClient" <100>

The server is not behind NAT, but the client side is. Any ideas on what I may be doing wrong? Thank you in advance for any help!

2

Enable the ICE support in your asterisk configuration.

3

Thank you. I just set up the ICE configuration but am still having the same issue. Here’s my rtp.conf:

[general] rtpstart=10000 rtpend=20000 icesupport=true stunaddr=stun.l.google.com:19302 turnaddr=numb.viagenie.ca turnusername=my-numb-email turnpassword=my-numb-password

ICE appears to be working based on the INVITE here, but still only ‘sent’ RTP packets with none received on the server, and no audio on the client. Asterisk SIP log:

<--- SIP read from WS:71.234.235.104:37029 --->
INVITE sip:602@162.243.116.82 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKRyQTc7z8LpJp86hJ8shBePastadLulk6;rport
From: "WebRTCClient"<sip:WebRTCClient@162.243.116.82>;tag=uTGPCkoXl6Fwd86cBc8S
To: <sip:602@162.243.116.82>
Contact: "WebRTCClient"<sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 1cab0f57-23da-48bb-8075-e92686fd1a24
CSeq: 39283 INVITE
Content-Type: application/sdp
Content-Length: 1830
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom

v=0
o=- 1289212349319112700 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS lRCvDDkAoTgxCra5YZPqiqhQRtsPrgaL95fG
m=audio 47098 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 71.234.235.104
a=rtcp:47098 IN IP4 71.234.235.104
a=candidate:1407141326 1 udp 2113937151 10.0.1.2 53425 typ host generation 0
a=candidate:1407141326 2 udp 2113937151 10.0.1.2 53425 typ host generation 0
a=candidate:1824104935 1 udp 1845501695 71.234.235.104 47098 typ srflx raddr 10.0.1.2 rport 53425 generation 0
a=candidate:1824104935 2 udp 1845501695 71.234.235.104 47098 typ srflx raddr 10.0.1.2 rport 53425 generation 0
a=candidate:492615998 1 tcp 1509957375 10.0.1.2 0 typ host generation 0
a=candidate:492615998 2 tcp 1509957375 10.0.1.2 0 typ host generation 0
a=ice-ufrag:rSfnj0EJ4kPaOvfk
a=ice-pwd:X5CYAzVLpNAHPh5pOLp7i4Ix
a=ice-options:google-ice
a=fingerprint:sha-256 9C:ED:BE:E4:29:C0:56:74:69:31:58:2F:C7:1D:BC:9D:08:86:DA:06:AE:E6:F1:AE:6E:34:C7:23:04:AE:99:AD
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:bVDOaCbo+AwsaHyXgi1ofEggt56m1ilxn6jdflZU
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7eHPDB9i8CoHblGOJJ5/sNR/OtQDd6km8+beVQyb
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3341534883 cname:FmVVh+9puW6TWdgX
a=ssrc:3341534883 msid:lRCvDDkAoTgxCra5YZPqiqhQRtsPrgaL95fG lRCvDDkAoTgxCra5YZPqiqhQRtsPrgaL95fGa0
a=ssrc:3341534883 mslabel:lRCvDDkAoTgxCra5YZPqiqhQRtsPrgaL95fG
a=ssrc:3341534883 label:lRCvDDkAoTgxCra5YZPqiqhQRtsPrgaL95fGa0
<------------->
--- (12 headers 41 lines) ---
Using INVITE request as basis request - 1cab0f57-23da-48bb-8075-e92686fd1a24
Found peer 'WebRTCClient' for 'WebRTCClient' from 71.234.235.104:37029
  == Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 71.234.235.104:47098
Looking for 602 in webrtc (domain 162.243.116.82)
list_route: hop: <sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>

<--- Transmitting (NAT) to 71.234.235.104:37029 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKRyQTc7z8LpJp86hJ8shBePastadLulk6;received=71.234.235.104;rport=37029
From: "WebRTCClient"<sip:WebRTCClient@162.243.116.82>;tag=uTGPCkoXl6Fwd86cBc8S
To: <sip:602@162.243.116.82>
Call-ID: 1cab0f57-23da-48bb-8075-e92686fd1a24
CSeq: 39283 INVITE
Server: Sipml5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:602@162.243.116.82:5060;transport=WS>
Content-Length: 0


<------------>
    -- Executing [602@webrtc:1] NoOp("SIP/WebRTCClient-00000000", "") in new stack
    -- Executing [602@webrtc:2] ConfBridge("SIP/WebRTCClient-00000000", "602") in new stack
Audio is at 10244
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 71.234.235.104:37029 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKRyQTc7z8LpJp86hJ8shBePastadLulk6;received=71.234.235.104;rport=37029
From: "WebRTCClient"<sip:WebRTCClient@162.243.116.82>;tag=uTGPCkoXl6Fwd86cBc8S
To: <sip:602@162.243.116.82>;tag=as015bc993
Call-ID: 1cab0f57-23da-48bb-8075-e92686fd1a24
CSeq: 39283 INVITE
Server: Sipml5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:602@162.243.116.82:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 380

v=0
o=root 1543123581 1543123581 IN IP4 162.243.116.82
s=Asterisk PBX 11.7.0-rc1
c=IN IP4 162.243.116.82
t=0 0
m=audio 10244 RTP/SAVPF 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:2FyKpgrEBSY1AIC+EIPhDVK9INb711R9MjwHy6jS

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

<--- SIP read from WS:71.234.235.104:37029 --->
ACK sip:602@162.243.116.82:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKyD86IoUwfpWKPPeKnXPf;rport
From: "WebRTCClient"<sip:WebRTCClient@162.243.116.82>;tag=uTGPCkoXl6Fwd86cBc8S
To: <sip:602@162.243.116.82>;tag=as015bc993
Contact: "WebRTCClient"<sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 1cab0f57-23da-48bb-8075-e92686fd1a24
CSeq: 39283 ACK
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom

<------------->
--- (11 headers 0 lines) ---
    -- <SIP/WebRTCClient-00000000> Playing 'conf-onlyperson.gsm' (language 'en')
    -- <SIP/WebRTCClient-00000000> Playing 'confbridge-join.gsm' (language 'en')
    -- <Bridge/0x7f818c0056e8-input> Playing 'confbridge-join.gsm' (language 'en')
Really destroying SIP dialog '52c38dec-c22d-497a-51bb-49eebfdb3aa5' Method: REGISTER
4

Also in the peer settings(sip.conf), when you enable that reload the config and enable the RTP debug, check for the (via ICE) in the rtp debug.

Check the IP address used in the SDP for the audio stream and compare with the RTP debug.

5

ICE is configured on both the client and server. Still the same issue though - no audio. Here’s what I’m seeing now:

Sent RTP packet to      66.228.45.110:58242 (type 00, seq 003090, ts 4537448, len 000164)
Sent RTP packet to      66.228.45.110:58242 (type 00, seq 003091, ts 4537608, len 000164)
Sent RTP packet to      66.228.45.110:58242 (type 00, seq 003092, ts 4537768, len 000164)
Sent RTP packet to      66.228.45.110:58242 (type 00, seq 003093, ts 4537928, len 000164)
Sent RTP packet to      66.228.45.110:58242 (type 00, seq 003094, ts 4538088, len 000164)
Sent RTP packet to      66.228.45.110:58242 (type 00, seq 003095, ts 4538248, len 000164)
Sent RTP packet to      66.228.45.110:58242 (type 00, seq 003096, ts 4538408, len 000164)

The IP above is listed in the SDP and is the address of the TURN server I’m using (numb). Should it be saying (via ICE) in this case?

Here’s the SIP output on Asterisk:

<--- SIP read from WS:[client-ip]:42583 --->
REGISTER sip:[asterisk-server-ip] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKndnTmfDdch2nT5iEop4gautWeo01Pc8x;rport
From: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=3mthTCGT07ws4q9VtORB
To: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>
Contact: "WebRTCClient"<sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7d397b10-6596-69ea-c18b-404520cb17a8
CSeq: 38853 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom
Supported: path

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

<--- Transmitting (no NAT) to [client-ip]:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKndnTmfDdch2nT5iEop4gautWeo01Pc8x;rport;received=[client-ip]
From: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=3mthTCGT07ws4q9VtORB
To: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=as17447c52
Call-ID: 7d397b10-6596-69ea-c18b-404520cb17a8
CSeq: 38853 REGISTER
Server: Sipml5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="[asterisk-server-ip]", nonce="6c5df958"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7d397b10-6596-69ea-c18b-404520cb17a8' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:[client-ip]:42583 --->
REGISTER sip:[asterisk-server-ip] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKrSwe2sTHmiJdEcnTZc2TNmU4Jn5N01Mg;rport
From: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=3mthTCGT07ws4q9VtORB
To: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>
Contact: "WebRTCClient"<sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 7d397b10-6596-69ea-c18b-404520cb17a8
CSeq: 38854 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="WebRTCClient",realm="[asterisk-server-ip]",nonce="6c5df958",uri="sip:[asterisk-server-ip]",response="fb66f849702fb27d8562e77a080ab2eb",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom
Supported: path

<------------->
--- (13 headers 0 lines) ---
    -- Registered SIP 'WebRTCClient' at [client-ip]:42583

<--- Transmitting (no NAT) to [client-ip]:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKrSwe2sTHmiJdEcnTZc2TNmU4Jn5N01Mg;rport;received=[client-ip]
From: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=3mthTCGT07ws4q9VtORB
To: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=as17447c52
Call-ID: 7d397b10-6596-69ea-c18b-404520cb17a8
CSeq: 38854 REGISTER
Server: Sipml5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 200
Contact: <sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws>;expires=200
Date: Tue, 10 Dec 2013 14:36:52 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7d397b10-6596-69ea-c18b-404520cb17a8' in 32000 ms (Method: REGISTER)

<--- SIP read from WS:[client-ip]:42583 --->
INVITE sip:602@[asterisk-server-ip] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKN3Nh8fdbLyCURyPTC4NDpflkRZHDmJaJ;rport
From: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=xLTRQ5ZaLiLkRs3pCRka
To: <sip:602@[asterisk-server-ip]>
Contact: "WebRTCClient"<sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 778d2a40-728f-770c-441b-bca0287669b0
CSeq: 42401 INVITE
Content-Type: application/sdp
Content-Length: 2056
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom

v=0
o=- 7786388717537124000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS uPB4yf7ndeFzJKdVp8d2GXQvLhtkpZRfVs40
m=audio 63068 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 [turn-server-ip]
a=rtcp:63068 IN IP4 [turn-server-ip]
a=candidate:1407141326 1 udp 2113937151 10.0.1.2 59055 typ host generation 0
a=candidate:1407141326 2 udp 2113937151 10.0.1.2 59055 typ host generation 0
a=candidate:1824104935 1 udp 1845501695 [client-ip] 48836 typ srflx raddr 10.0.1.2 rport 59055 generation 0
a=candidate:1824104935 2 udp 1845501695 [client-ip] 48836 typ srflx raddr 10.0.1.2 rport 59055 generation 0
a=candidate:492615998 1 tcp 1509957375 10.0.1.2 0 typ host generation 0
a=candidate:492615998 2 tcp 1509957375 10.0.1.2 0 typ host generation 0
a=candidate:811261687 1 udp 33562367 [turn-server-ip] 63068 typ relay raddr [client-ip] rport 39121 generation 0
a=candidate:811261687 2 udp 33562367 [turn-server-ip] 63068 typ relay raddr [client-ip] rport 39121 generation 0
a=ice-ufrag:d+a8dBfjq4lPuaeU
a=ice-pwd:sXwFTq7sD3KB/DWyqnEd1woL
a=ice-options:google-ice
a=fingerprint:sha-256 9C:ED:BE:E4:29:C0:56:74:69:31:58:2F:C7:1D:BC:9D:08:86:DA:06:AE:E6:F1:AE:6E:34:C7:23:04:AE:99:AD
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:mdAeRcv/2ZRgRDJ9MbY1rRyJuMGa4d/nHipMeiTu
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HweUvGCM5DhugcRkHlegualG10yWKgjO+ksT7ibJ
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2227985394 cname:MjtcprLm+xRQ5CaR
a=ssrc:2227985394 msid:uPB4yf7ndeFzJKdVp8d2GXQvLhtkpZRfVs40 uPB4yf7ndeFzJKdVp8d2GXQvLhtkpZRfVs40a0
a=ssrc:2227985394 mslabel:uPB4yf7ndeFzJKdVp8d2GXQvLhtkpZRfVs40
a=ssrc:2227985394 label:uPB4yf7ndeFzJKdVp8d2GXQvLhtkpZRfVs40a0
<------------->
--- (12 headers 43 lines) ---
Using INVITE request as basis request - 778d2a40-728f-770c-441b-bca0287669b0
Found peer 'WebRTCClient' for 'WebRTCClient' from [client-ip]:42583

<--- Reliably Transmitting (no NAT) to [client-ip]:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKN3Nh8fdbLyCURyPTC4NDpflkRZHDmJaJ;rport;received=[client-ip]
From: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=xLTRQ5ZaLiLkRs3pCRka
To: <sip:602@[asterisk-server-ip]>;tag=as650e4948
Call-ID: 778d2a40-728f-770c-441b-bca0287669b0
CSeq: 42401 INVITE
Server: Sipml5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="[asterisk-server-ip]", nonce="6be7afaf"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '778d2a40-728f-770c-441b-bca0287669b0' in 32000 ms (Method: INVITE)

<--- SIP read from WS:[client-ip]:42583 --->
ACK sip:602@[asterisk-server-ip] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKN3Nh8fdbLyCURyPTC4NDpflkRZHDmJaJ;rport
From: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=xLTRQ5ZaLiLkRs3pCRka
To: <sip:602@[asterisk-server-ip]>;tag=as650e4948
Call-ID: 778d2a40-728f-770c-441b-bca0287669b0
CSeq: 42401 ACK
Content-Length: 0
Max-Forwards: 70

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

<--- SIP read from WS:[client-ip]:42583 --->
INVITE sip:602@[asterisk-server-ip] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKnTQTWp32CFGBoywZSmLMqcmclsWd2QmV;rport
From: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=xLTRQ5ZaLiLkRs3pCRka
To: <sip:602@[asterisk-server-ip]>
Contact: "WebRTCClient"<sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 778d2a40-728f-770c-441b-bca0287669b0
CSeq: 42402 INVITE
Content-Type: application/sdp
Content-Length: 2056
Max-Forwards: 70
Authorization: Digest username="WebRTCClient",realm="[asterisk-server-ip]",nonce="6be7afaf",uri="sip:602@[asterisk-server-ip]",response="4c2cf3aed2626963671bfdddfd8ecbed",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom

v=0
o=- 7786388717537124000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS uPB4yf7ndeFzJKdVp8d2GXQvLhtkpZRfVs40
m=audio 63068 RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 [turn-server-ip]
a=rtcp:63068 IN IP4 [turn-server-ip]
a=candidate:1407141326 1 udp 2113937151 10.0.1.2 59055 typ host generation 0
a=candidate:1407141326 2 udp 2113937151 10.0.1.2 59055 typ host generation 0
a=candidate:1824104935 1 udp 1845501695 [client-ip] 48836 typ srflx raddr 10.0.1.2 rport 59055 generation 0
a=candidate:1824104935 2 udp 1845501695 [client-ip] 48836 typ srflx raddr 10.0.1.2 rport 59055 generation 0
a=candidate:492615998 1 tcp 1509957375 10.0.1.2 0 typ host generation 0
a=candidate:492615998 2 tcp 1509957375 10.0.1.2 0 typ host generation 0
a=candidate:811261687 1 udp 33562367 [turn-server-ip] 63068 typ relay raddr [client-ip] rport 39121 generation 0
a=candidate:811261687 2 udp 33562367 [turn-server-ip] 63068 typ relay raddr [client-ip] rport 39121 generation 0
a=ice-ufrag:d+a8dBfjq4lPuaeU
a=ice-pwd:sXwFTq7sD3KB/DWyqnEd1woL
a=ice-options:google-ice
a=fingerprint:sha-256 9C:ED:BE:E4:29:C0:56:74:69:31:58:2F:C7:1D:BC:9D:08:86:DA:06:AE:E6:F1:AE:6E:34:C7:23:04:AE:99:AD
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:mdAeRcv/2ZRgRDJ9MbY1rRyJuMGa4d/nHipMeiTu
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HweUvGCM5DhugcRkHlegualG10yWKgjO+ksT7ibJ
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2227985394 cname:MjtcprLm+xRQ5CaR
a=ssrc:2227985394 msid:uPB4yf7ndeFzJKdVp8d2GXQvLhtkpZRfVs40 uPB4yf7ndeFzJKdVp8d2GXQvLhtkpZRfVs40a0
a=ssrc:2227985394 mslabel:uPB4yf7ndeFzJKdVp8d2GXQvLhtkpZRfVs40
a=ssrc:2227985394 label:uPB4yf7ndeFzJKdVp8d2GXQvLhtkpZRfVs40a0
<------------->
--- (13 headers 43 lines) ---
Using INVITE request as basis request - 778d2a40-728f-770c-441b-bca0287669b0
Found peer 'WebRTCClient' for 'WebRTCClient' from [client-ip]:42583
  == Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found unknown media description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port [turn-server-ip]:63068
Looking for 602 in webrtc (domain [asterisk-server-ip])
list_route: hop: <sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>

<--- Transmitting (no NAT) to [client-ip]:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKnTQTWp32CFGBoywZSmLMqcmclsWd2QmV;rport;received=[client-ip]
From: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=xLTRQ5ZaLiLkRs3pCRka
To: <sip:602@[asterisk-server-ip]>
Call-ID: 778d2a40-728f-770c-441b-bca0287669b0
CSeq: 42402 INVITE
Server: Sipml5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:602@[asterisk-server-ip]:5060;transport=WS>
Content-Length: 0


<------------>
    -- Executing [602@webrtc:1] NoOp("SIP/WebRTCClient-00000000", "") in new stack
    -- Executing [602@webrtc:2] ConfBridge("SIP/WebRTCClient-00000000", "602") in new stack
Audio is at 7890
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to [client-ip]:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKnTQTWp32CFGBoywZSmLMqcmclsWd2QmV;rport;received=[client-ip]
From: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=xLTRQ5ZaLiLkRs3pCRka
To: <sip:602@[asterisk-server-ip]>;tag=as08f7d1e3
Call-ID: 778d2a40-728f-770c-441b-bca0287669b0
CSeq: 42402 INVITE
Server: Sipml5
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:602@[asterisk-server-ip]:5060;transport=WS>
Content-Type: application/sdp
Content-Length: 379

v=0
o=root 1798667405 1798667405 IN IP4 [asterisk-server-ip]
s=Asterisk PBX 11.7.0-rc1
c=IN IP4 [asterisk-server-ip]
t=0 0
m=audio 7890 RTP/SAVPF 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:RLFDPt3RwTDRiXX9eh+LVx+KN7FP1ybC+oTxokuX

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

<--- SIP read from WS:[client-ip]:42583 --->
ACK sip:602@[asterisk-server-ip]:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK6EdGyVuAmw9zrs2UWu5T;rport
From: "WebRTCClient"<sip:WebRTCClient@[asterisk-server-ip]>;tag=xLTRQ5ZaLiLkRs3pCRka
To: <sip:602@[asterisk-server-ip]>;tag=as08f7d1e3
Contact: "WebRTCClient"<sip:WebRTCClient@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 778d2a40-728f-770c-441b-bca0287669b0
CSeq: 42402 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="WebRTCClient",realm="[asterisk-server-ip]",nonce="6be7afaf",uri="sip:602@[asterisk-server-ip]:5060;transport=WS",response="3577262cadb8c51f7257b01f204be578",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom

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

And my latest sip.conf:

[general]
realm=w.x.y.z
udpbindaddr=0.0.0.0:5060
transport=udp,ws,wss
useragent=Sipml5
icesupport = yes

[WebRTCClient]
type=friend
host=dynamic
context=webrtc
hasiax = no
hassip = yes
encryption = yes
avpf = yes
icesupport = yes
videosupport=no
directmedia=no

Any ideas? Thanks in advance!

6

Seems like your asterisk server isn’t compiled with ICE support, try to install uuid-devel and libuuid-devel then recompile asterisk.

7

I installed uuid-dev (on ubuntu) and recompiled and all is well now. Thank you so much for helping me track this down!

8

Okay, next problem. Maybe I should create a new thread for this, but hopefully it’s an easy fix.

I have two clients connected to Asterisk from the same computer. Things are working great for a few minutes, then the audio goes away. RTP logs show this at the cutoff point:

Sent RTP packet to      66.228.45.110:63533 (via ICE) (type 00, seq 063563, ts 2410888, len 4294967284)
Sent RTP packet to      66.228.45.110:63532 (via ICE) (type 00, seq 059131, ts 2476824, len 4294967284)
Got  RTP packet from    66.228.45.110:63533 (type 00, seq 036913, ts 2079594190, len 000160)
Sent RTP packet to      66.228.45.110:63532 (via ICE) (type 00, seq 059132, ts 2476984, len 4294967284)
Sent RTP packet to      66.228.45.110:63533 (via ICE) (type 00, seq 063564, ts 2411048, len 4294967284)
Got  RTP packet from    66.228.45.110:63533 (type 00, seq 036914, ts 2079594350, len 000160)
Sent RTP packet to      66.228.45.110:63532 (via ICE) (type 00, seq 059133, ts 2477144, len 4294967284)
Sent RTP packet to      66.228.45.110:63533 (via ICE) (type 00, seq 063565, ts 2411208, len 4294967284)
Got  RTP packet from    66.228.45.110:63533 (type 00, seq 036915, ts 2079594510, len 000160)
Sent RTP packet to      66.228.45.110:63532 (via ICE) (type 00, seq 059134, ts 2477304, len 4294967284)
Sent RTP packet to      66.228.45.110:63533 (via ICE) (type 00, seq 063566, ts 2411368, len 4294967284)
Got  RTP packet from    66.228.45.110:63533 (type 00, seq 036916, ts 2079594670, len 000160)
Sent RTP packet to      66.228.45.110:63532 (via ICE) (type 00, seq 059135, ts 2477464, len 4294967284)
Sent RTP packet to      66.228.45.110:63533 (via ICE) (type 00, seq 063567, ts 2411528, len 4294967284)
Got  RTP packet from    66.228.45.110:63533 (type 00, seq 036917, ts 2079594830, len 000160)
Sent RTP packet to      66.228.45.110:63532 (via ICE) (type 00, seq 059136, ts 2477624, len 4294967284)
Sent RTP packet to      66.228.45.110:63533 (via ICE) (type 00, seq 063568, ts 2411688, len 4294967284)
Sent RTP packet to      66.228.45.110:63532 (via ICE) (type 00, seq 059137, ts 2477784, len 4294967284)
Sent RTP packet to      66.228.45.110:63533 (via ICE) (type 00, seq 063569, ts 2411848, len 4294967284)
Sent RTP packet to      66.228.45.110:63532 (via ICE) (type 00, seq 059138, ts 2477944, len 4294967284)
Sent RTP packet to      66.228.45.110:63533 (via ICE) (type 00, seq 063570, ts 2412008, len 4294967284)
Sent RTP packet to      66.228.45.110:63532 (via ICE) (type 00, seq 059139, ts 2478104, len 4294967284)
Sent RTP packet to      66.228.45.110:63533 (via ICE) (type 00, seq 063571, ts 2412168, len 4294967284)
Sent RTP packet to      66.228.45.110:63533 (via ICE) (type 00, seq 063572, ts 2412328, len 4294967284)
Sent RTP packet to      66.228.45.110:63532 (via ICE) (type 00, seq 059140, ts 2478264, len 4294967284)
Sent RTP packet to      66.228.45.110:63533 (via ICE) (type 00, seq 063573, ts 2412488, len 4294967284)
Sent RTP packet to      66.228.45.110:63532 (via ICE) (type 00, seq 059141, ts 2478424, len 4294967284)

I enabled my firewall for testing so this is all going through the TURN server.

9

If you are in the same lan of the server the RTP is sending to the public IP instead the local ip, if not and the Public IP is the correct check with a pcap trace what is happening or check the NAT settings for the sip peer.