WebRTC no sound

Hi,
I know that this is a problem that has been discussed here multiple times. I’ve went over most of the threads, including the two stickied ones. I’d still however like some pointers as to what could be wrong in my configuration. Everything works fine if I use some softphone software (like Zoiper), but I’m having trouble with WebRTC. WebRTC only works when I connect from LAN. Otherwise, the call is started, but there is no sound.

My server is behind a NAT, but all of the ports needed (5060, 8088 TCP, 10000-20000 UDP) are forwarded to the server.

I’m running Asterisk 11.16.0 on Debian.

I reinstalled everything and tried with a fresh configuration (copied from one of the stickied threads). Just in case, here is my sip.conf:

[general]

context=guest
transport=udp,ws
rtcachefriends=yes
allowguest=yes
limitonpeers=yes
callcounter=yes
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0:5060
externip=[SERVER_IP]
externrefresh=150
localnet=192.168.0.0/255.255.255.0
disallow=all
;allow=g729                   ; First disallow all codecs
allow=gsm
allow=ulaw                     ; Allow codecs in order of preference
allow=alaw                     ; Allow codecs in order of preference
language=en                    ; Default language setting for all users/peers
callcounter=yes
limitonpeers=yes
callevents=yes
useragent=Digital-Merge_UA
realm=[SERVER_IP]
nat=force_rport,comedia

[5005]
type=friend
secret=randpass
host=dynamic
context=wrtc
disallow=all
allow=ulaw
allow=alaw
;allow=g729
;allow=gsm
;allow=h263p
;allow=h264
dtmf=auto
;videosupport=yes
transport=ws,udp
avpf=yes
nat=force_rport,comedia
callerid="WebRTC"<5005>
encryption=yes
qualify=yes
icesupport=yes
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS

SIP Debug:

[code]<— SIP read from WS:[CLIENT_IP]:54959 —>
INVITE sip:1000@[SERVER_IP] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK31HnvvVGsQTo4Q0Z8AXdStr3V3fQvrPp;rport
From: "5005"sip:5005@SERVER_IP;tag=zlLLQ4QNWLKJcttTRTDv
To: sip:1000@SERVER_IP
Contact: "5005"sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;language=“en,fr”
Call-ID: d77ad2b3-a2b3-1326-9613-07620ead0772
CSeq: 30631 INVITE
Content-Type: application/sdp
Content-Length: 1827
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 1335183969167303700 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 3iZ9z3iFxDIECifQCiOAchRm84Tjl9jEBjG6
m=audio 54974 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 [CLIENT_IP]
a=rtcp:54974 IN IP4 [CLIENT_IP]
a=candidate:3264781818 1 udp 2122260223 192.168.137.85 56967 typ host generation 0
a=candidate:3264781818 2 udp 2122260223 192.168.137.85 56967 typ host generation 0
a=candidate:2350604554 1 tcp 1518280447 192.168.137.85 0 typ host tcptype active generation 0
a=candidate:2350604554 2 tcp 1518280447 192.168.137.85 0 typ host tcptype active generation 0
a=candidate:236298857 1 udp 1686052607 [CLIENT_IP] 54974 typ srflx raddr 192.168.137.85 rport 56967 generation 0
a=candidate:236298857 2 udp 1686052607 [CLIENT_IP] 54974 typ srflx raddr 192.168.137.85 rport 56967 generation 0
a=ice-ufrag:I8rM/inRGGyEmU1j
a=ice-pwd:38TIsWEomSUtexsfj8pHE+i4
a=ice-options:google-ice
a=fingerprint:sha-256 09:FC:7D:C5:89:BE:4A:9F:43:AB:F8:F7:78:60:AE:DA:34:A5:98:D5:98:55:FF:E9:04:7B:AF:04:18:A8:6F:FF
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
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:3888884909 cname:v66S537/4Z4Nx3EN
a=ssrc:3888884909 msid:3iZ9z3iFxDIECifQCiOAchRm84Tjl9jEBjG6 58c96f72-8a22-4954-a7de-321e431c40ae
a=ssrc:3888884909 mslabel:3iZ9z3iFxDIECifQCiOAchRm84Tjl9jEBjG6
a=ssrc:3888884909 label:58c96f72-8a22-4954-a7de-321e431c40ae
<------------->
— (12 headers 41 lines) —
Using INVITE request as basis request - d77ad2b3-a2b3-1326-9613-07620ead0772
Found peer ‘5005’ for ‘5005’ from [CLIENT_IP]:54959

<— Reliably Transmitting (NAT) to [CLIENT_IP]:54959 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK31HnvvVGsQTo4Q0Z8AXdStr3V3fQvrPp;received=[CLIENT_IP];rport=54959
From: “5005"sip:5005@SERVER_IP;tag=zlLLQ4QNWLKJcttTRTDv
To: sip:1000@SERVER_IP;tag=as5acd0f7c
Call-ID: d77ad2b3-a2b3-1326-9613-07620ead0772
CSeq: 30631 INVITE
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=”[SERVER_IP]", nonce=“4a517fcc”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘d77ad2b3-a2b3-1326-9613-07620ead0772’ in 6400 ms (Method: INVITE)

<— SIP read from WS:[CLIENT_IP]:54959 —>
ACK sip:1000@[SERVER_IP] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK31HnvvVGsQTo4Q0Z8AXdStr3V3fQvrPp;rport
From: "5005"sip:5005@SERVER_IP;tag=zlLLQ4QNWLKJcttTRTDv
To: sip:1000@SERVER_IP;tag=as5acd0f7c
Call-ID: d77ad2b3-a2b3-1326-9613-07620ead0772
CSeq: 30631 ACK
Content-Length: 0
Max-Forwards: 70

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

<— SIP read from WS:[CLIENT_IP]:54959 —>
INVITE sip:1000@[SERVER_IP] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcTVGAs6ea1qMqJzXnp4o7eCxacLDPuST;rport
From: “5005"sip:5005@SERVER_IP;tag=zlLLQ4QNWLKJcttTRTDv
To: sip:1000@SERVER_IP
Contact: “5005"sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;language=“en,fr”
Call-ID: d77ad2b3-a2b3-1326-9613-07620ead0772
CSeq: 30632 INVITE
Content-Type: application/sdp
Content-Length: 1827
Max-Forwards: 70
Authorization: Digest username=“5005”,realm=”[SERVER_IP]”,nonce=“4a517fcc”,uri=“sip:1000@[SERVER_IP]”,response=“78e9f793a3baddc93f58ee39cfd5f843”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 1335183969167303700 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 3iZ9z3iFxDIECifQCiOAchRm84Tjl9jEBjG6
m=audio 54974 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 [CLIENT_IP]
a=rtcp:54974 IN IP4 [CLIENT_IP]
a=candidate:3264781818 1 udp 2122260223 192.168.137.85 56967 typ host generation 0
a=candidate:3264781818 2 udp 2122260223 192.168.137.85 56967 typ host generation 0
a=candidate:2350604554 1 tcp 1518280447 192.168.137.85 0 typ host tcptype active generation 0
a=candidate:2350604554 2 tcp 1518280447 192.168.137.85 0 typ host tcptype active generation 0
a=candidate:236298857 1 udp 1686052607 [CLIENT_IP] 54974 typ srflx raddr 192.168.137.85 rport 56967 generation 0
a=candidate:236298857 2 udp 1686052607 [CLIENT_IP] 54974 typ srflx raddr 192.168.137.85 rport 56967 generation 0
a=ice-ufrag:I8rM/inRGGyEmU1j
a=ice-pwd:38TIsWEomSUtexsfj8pHE+i4
a=ice-options:google-ice
a=fingerprint:sha-256 09:FC:7D:C5:89:BE:4A:9F:43:AB:F8:F7:78:60:AE:DA:34:A5:98:D5:98:55:FF:E9:04:7B:AF:04:18:A8:6F:FF
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
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:3888884909 cname:v66S537/4Z4Nx3EN
a=ssrc:3888884909 msid:3iZ9z3iFxDIECifQCiOAchRm84Tjl9jEBjG6 58c96f72-8a22-4954-a7de-321e431c40ae
a=ssrc:3888884909 mslabel:3iZ9z3iFxDIECifQCiOAchRm84Tjl9jEBjG6
a=ssrc:3888884909 label:58c96f72-8a22-4954-a7de-321e431c40ae
<------------->
— (13 headers 41 lines) —
Using INVITE request as basis request - d77ad2b3-a2b3-1326-9613-07620ead0772
Found peer ‘5005’ for ‘5005’ from [CLIENT_IP]:54959
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
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 G722 for ID 9
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 - (ulaw|alaw), peer - audio=(ulaw|alaw|g722)/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 [CLIENT_IP]:54974
Looking for 1000 in wrtc (domain [SERVER_IP])
list_route: hop: sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws

<— Transmitting (NAT) to [CLIENT_IP]:54959 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcTVGAs6ea1qMqJzXnp4o7eCxacLDPuST;received=[CLIENT_IP];rport=54959
From: "5005"sip:5005@SERVER_IP;tag=zlLLQ4QNWLKJcttTRTDv
To: sip:1000@SERVER_IP
Call-ID: d77ad2b3-a2b3-1326-9613-07620ead0772
CSeq: 30632 INVITE
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1000@SERVER_IP:0;transport=WS
Content-Length: 0

<------------>
Audio is at 19482
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to [CLIENT_IP]:54959 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcTVGAs6ea1qMqJzXnp4o7eCxacLDPuST;received=[CLIENT_IP];rport=54959
From: "5005"sip:5005@SERVER_IP;tag=zlLLQ4QNWLKJcttTRTDv
To: sip:1000@SERVER_IP;tag=as5035a2d0
Call-ID: d77ad2b3-a2b3-1326-9613-07620ead0772
CSeq: 30632 INVITE
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:1000@SERVER_IP:0;transport=WS
Content-Type: application/sdp
Content-Length: 851

v=0
o=root 197032155 197032155 IN IP4 [SERVER_IP]
s=Asterisk PBX 11.16.0
c=IN IP4 [SERVER_IP]
t=0 0
m=audio 19482 UDP/TLS/RTP/SAVPF 0 8 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=ice-ufrag:23f0e9ed7ffcebce34b95448714d4c12
a=ice-pwd:60ecf3eb287f00e943ccea5c4320c625
a=candidate:Hc0a80146 1 UDP 2130706431 192.168.1.70 19482 typ host
a=candidate:S5abe665e 1 UDP 1694498815 [SERVER_IP] 54896 typ srflx raddr 192.168.1.70 rport 19482
a=candidate:Hc0a80146 2 UDP 2130706430 192.168.1.70 19483 typ host
a=candidate:S5abe665e 2 UDP 1694498814 [SERVER_IP] 64266 typ srflx raddr 192.168.1.70 rport 19483
a=connection:new
a=setup:active
a=fingerprint:SHA-256 F8:47:D2:58:1B:90:C6:91:28:3A:99:FA:1D:D5:08:2F:3E:80:09:B1:9E:6A:FC:D5:56:F5:90:AB:D9:2E:58:E6
a=sendrecv

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

<— SIP read from WS:[CLIENT_IP]:54959 —>
ACK sip:1000@[SERVER_IP];transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK9r6h9l8eAdataMTQo7RO;rport
From: “5005"sip:5005@SERVER_IP;tag=zlLLQ4QNWLKJcttTRTDv
To: sip:1000@SERVER_IP;tag=as5035a2d0
Contact: “5005"sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;language=“en,fr”
Call-ID: d77ad2b3-a2b3-1326-9613-07620ead0772
CSeq: 30632 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“5005”,realm=”[SERVER_IP]”,nonce=“4a517fcc”,uri=“sip:1000@[SERVER_IP];transport=WS”,response=“473c43ba2d72e8481b55473e5474de45”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

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

<— SIP read from WS:[CLIENT_IP]:54959 —>
BYE sip:1000@[SERVER_IP];transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKJgaS8PPkv4SDhFPqbEH85HTfCD5x7URE;rport
From: “5005"sip:5005@SERVER_IP;tag=zlLLQ4QNWLKJcttTRTDv
To: sip:1000@SERVER_IP;tag=as5035a2d0
Call-ID: d77ad2b3-a2b3-1326-9613-07620ead0772
CSeq: 30633 BYE
Content-Length: 0
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language=“en,fr”
Authorization: Digest username=“5005”,realm=”[SERVER_IP]",nonce=“4a517fcc”,uri=“sip:1000@[SERVER_IP];transport=WS”,response=“6b1765578c7d967111a38eced1d67077”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

<------------->
— (13 headers 0 lines) —
Scheduling destruction of SIP dialog ‘d77ad2b3-a2b3-1326-9613-07620ead0772’ in 6400 ms (Method: BYE)

<— Transmitting (NAT) to [CLIENT_IP]:54959 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKJgaS8PPkv4SDhFPqbEH85HTfCD5x7URE;received=[CLIENT_IP];rport=54959
From: "5005"sip:5005@SERVER_IP;tag=zlLLQ4QNWLKJcttTRTDv
To: sip:1000@SERVER_IP;tag=as5035a2d0
Call-ID: d77ad2b3-a2b3-1326-9613-07620ead0772
CSeq: 30633 BYE
Server: Digital-Merge_UA
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
[/code]

JS Debug (*note: not from the same session as the SIP debug log, so the logs do not add up 1:1):

[code]SIPml-api.js?svn=222:1 State machine: c0000_Started_2_Outgoing_X_oINVITE
SIPml-api.js?svn=222:1 PeerConnectionClass = function RTCPeerConnection() { [native code] } SessionDescriptionClass = function RTCSessionDescription() { [native code] } IceCandidateClass = function RTCIceCandidate() { [native code] }
SIPml-api.js?svn=222:1 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=222:1 ==stack event = m_permission_requested
SIPml-api.js?svn=222:1 ==session event = connecting
SIPml-api.js?svn=222:1 onGetUserMediaSuccess
SIPml-api.js?svn=222:1 createOffer
SIPml-api.js?svn=222:1 ==stack event = m_permission_accepted
SIPml-api.js?svn=222:1 ==session event = m_stream_audio_local_added
SIPml-api.js?svn=222:1 onNegotiationNeeded
SIPml-api.js?svn=222:1 onCreateSdpSuccess
SIPml-api.js?svn=222:1 onSetLocalDescriptionSuccess
SIPml-api.js?svn=222:1 onSignalingstateChange:have-local-offer
10SIPml-api.js?svn=222:1 onIceCandidate = gathering
SIPml-api.js?svn=222:1 onIceCandidate = complete
SIPml-api.js?svn=222:1 ICE GATHERING COMPLETED!
SIPml-api.js?svn=222:1 onIceGatheringCompleted
SIPml-api.js?svn=222:1 SEND: INVITE sip:1000@[SERVER_IP] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEtFLcZLk40gzo346O8QHnLCTy342QpCJ;rport
From: "5005"sip:5005@SERVER_IP;tag=dshEiqzXzTrTUo9auLNq
To: sip:1000@SERVER_IP
Contact: "5005"sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;language=“en,fr”
Call-ID: be92af04-2b40-de63-94f6-7bba1cf8f3a5
CSeq: 18803 INVITE
Content-Type: application/sdp
Content-Length: 2302
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 320749111154457800 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS XHt7LSBoUsYHvRdij21l9jmFRp3knXVnlU1I
m=audio 54704 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 [CLIENT_IP]
a=rtcp:54704 IN IP4 [CLIENT_IP]
a=candidate:3264781818 1 udp 2122260223 192.168.137.85 55157 typ host generation 0
a=candidate:3264781818 2 udp 2122260223 192.168.137.85 55157 typ host generation 0
a=candidate:2350604554 1 tcp 1518280447 192.168.137.85 0 typ host tcptype active generation 0
a=candidate:2350604554 2 tcp 1518280447 192.168.137.85 0 typ host tcptype active generation 0
a=candidate:236298857 1 udp 1686052607 [CLIENT_IP] 54704 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=candidate:236298857 2 udp 1686052607 [CLIENT_IP] 54704 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=candidate:236298857 1 udp 1686052607 [CLIENT_IP] 54703 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=candidate:236298857 2 udp 1686052607 [CLIENT_IP] 54703 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=candidate:236298857 1 udp 1686052607 [CLIENT_IP] 54705 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=candidate:236298857 2 udp 1686052607 [CLIENT_IP] 54705 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=ice-ufrag:jzyV1r4xyrOGef4a
a=ice-pwd:/xmCfaq8qaMlClfM9tPm2iJE
a=ice-options:google-ice
a=fingerprint:sha-256 09:FC:7D:C5:89:BE:4A:9F:43:AB:F8:F7:78:60:AE:DA:34:A5:98:D5:98:55:FF:E9:04:7B:AF:04:18:A8:6F:FF
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
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:2436512620 cname:nEQB9VuM/lA9uKxd
a=ssrc:2436512620 msid:XHt7LSBoUsYHvRdij21l9jmFRp3knXVnlU1I 857754a2-2f1f-49fd-9b38-3361dab7f1c3
a=ssrc:2436512620 mslabel:XHt7LSBoUsYHvRdij21l9jmFRp3knXVnlU1I
a=ssrc:2436512620 label:857754a2-2f1f-49fd-9b38-3361dab7f1c3

SIPml-api.js?svn=222:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=222:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=54702;received=[CLIENT_IP];branch=z9hG4bKEtFLcZLk40gzo346O8QHnLCTy342QpCJ
From: “5005"sip:5005@SERVER_IP;tag=dshEiqzXzTrTUo9auLNq
To: sip:1000@SERVER_IP;tag=as433b9b18
Call-ID: be92af04-2b40-de63-94f6-7bba1cf8f3a5
CSeq: 18803 INVITE
Content-Length: 0
Server: Digital-Merge_UA
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm=”[SERVER_IP]",nonce=“7e5f517e”,stale=FALSE,algorithm=MD5

SIPml-api.js?svn=222:1 SEND: ACK sip:1000@[SERVER_IP] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKEtFLcZLk40gzo346O8QHnLCTy342QpCJ;rport
From: "5005"sip:5005@SERVER_IP;tag=dshEiqzXzTrTUo9auLNq
To: sip:1000@SERVER_IP;tag=as433b9b18
Call-ID: be92af04-2b40-de63-94f6-7bba1cf8f3a5
CSeq: 18803 ACK
Content-Length: 0
Max-Forwards: 70

SIPml-api.js?svn=222:1 State machine: x0000_Any_2_Any_X_i401_407_INVITE
SIPml-api.js?svn=222:1 SEND: INVITE sip:1000@[SERVER_IP] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKknWrKkv6QegNR8mkTZ95m0k1pbNzh5VJ;rport
From: “5005"sip:5005@SERVER_IP;tag=dshEiqzXzTrTUo9auLNq
To: sip:1000@SERVER_IP
Contact: “5005"sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;language=“en,fr”
Call-ID: be92af04-2b40-de63-94f6-7bba1cf8f3a5
CSeq: 18804 INVITE
Content-Type: application/sdp
Content-Length: 2302
Max-Forwards: 70
Authorization: Digest username=“5005”,realm=”[SERVER_IP]”,nonce=“7e5f517e”,uri=“sip:1000@[SERVER_IP]”,response=“52d1937b4f5851f7a147c2e2e0abcabc”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

v=0
o=- 320749111154457800 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS XHt7LSBoUsYHvRdij21l9jmFRp3knXVnlU1I
m=audio 54704 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 [CLIENT_IP]
a=rtcp:54704 IN IP4 [CLIENT_IP]
a=candidate:3264781818 1 udp 2122260223 192.168.137.85 55157 typ host generation 0
a=candidate:3264781818 2 udp 2122260223 192.168.137.85 55157 typ host generation 0
a=candidate:2350604554 1 tcp 1518280447 192.168.137.85 0 typ host tcptype active generation 0
a=candidate:2350604554 2 tcp 1518280447 192.168.137.85 0 typ host tcptype active generation 0
a=candidate:236298857 1 udp 1686052607 [CLIENT_IP] 54704 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=candidate:236298857 2 udp 1686052607 [CLIENT_IP] 54704 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=candidate:236298857 1 udp 1686052607 [CLIENT_IP] 54703 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=candidate:236298857 2 udp 1686052607 [CLIENT_IP] 54703 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=candidate:236298857 1 udp 1686052607 [CLIENT_IP] 54705 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=candidate:236298857 2 udp 1686052607 [CLIENT_IP] 54705 typ srflx raddr 192.168.137.85 rport 55157 generation 0
a=ice-ufrag:jzyV1r4xyrOGef4a
a=ice-pwd:/xmCfaq8qaMlClfM9tPm2iJE
a=ice-options:google-ice
a=fingerprint:sha-256 09:FC:7D:C5:89:BE:4A:9F:43:AB:F8:F7:78:60:AE:DA:34:A5:98:D5:98:55:FF:E9:04:7B:AF:04:18:A8:6F:FF
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
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:2436512620 cname:nEQB9VuM/lA9uKxd
a=ssrc:2436512620 msid:XHt7LSBoUsYHvRdij21l9jmFRp3knXVnlU1I 857754a2-2f1f-49fd-9b38-3361dab7f1c3
a=ssrc:2436512620 mslabel:XHt7LSBoUsYHvRdij21l9jmFRp3knXVnlU1I
a=ssrc:2436512620 label:857754a2-2f1f-49fd-9b38-3361dab7f1c3

SIPml-api.js?svn=222:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=222:1 recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=54702;received=[CLIENT_IP];branch=z9hG4bKknWrKkv6QegNR8mkTZ95m0k1pbNzh5VJ
From: "5005"sip:5005@SERVER_IP;tag=dshEiqzXzTrTUo9auLNq
To: sip:1000@SERVER_IP
Contact: sip:1000@SERVER_IP;transport=WS
Call-ID: be92af04-2b40-de63-94f6-7bba1cf8f3a5
CSeq: 18804 INVITE
Content-Length: 0
Server: Digital-Merge_UA
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

SIPml-api.js?svn=222:1 State machine: x0000_Any_2_Any_X_i1xx
SIPml-api.js?svn=222:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=222:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=54702;received=[CLIENT_IP];branch=z9hG4bKknWrKkv6QegNR8mkTZ95m0k1pbNzh5VJ
From: "5005"sip:5005@SERVER_IP;tag=dshEiqzXzTrTUo9auLNq
To: sip:1000@SERVER_IP;tag=as22a066e7
Contact: sip:1000@SERVER_IP;transport=WS
Call-ID: be92af04-2b40-de63-94f6-7bba1cf8f3a5
CSeq: 18804 INVITE
Content-Type: application/sdp
Content-Length: 853
Server: Digital-Merge_UA
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

v=0
o=root 2099092377 2099092377 IN IP4 [SERVER_IP]
s=Asterisk PBX 11.16.0
c=IN IP4 [SERVER_IP]
t=0 0
m=audio 19720 UDP/TLS/RTP/SAVPF 0 8 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=ice-ufrag:028dbf70534845ca3353a97758ae64f0
a=ice-pwd:6c3e90bb5769116406a4769f51a7b134
a=candidate:Hc0a80146 1 UDP 2130706431 192.168.1.70 19720 typ host
a=candidate:S5abe665e 1 UDP 1694498815 [SERVER_IP] 60284 typ srflx raddr 192.168.1.70 rport 19720
a=candidate:Hc0a80146 2 UDP 2130706430 192.168.1.70 19721 typ host
a=candidate:S5abe665e 2 UDP 1694498814 [SERVER_IP] 53182 typ srflx raddr 192.168.1.70 rport 19721
a=connection:new
a=setup:active
a=fingerprint:SHA-256 F8:47:D2:58:1B:90:C6:91:28:3A:99:FA:1D:D5:08:2F:3E:80:09:B1:9E:6A:FC:D5:56:F5:90:AB:D9:2E:58:E6
a=sendrecv

SIPml-api.js?svn=222:1 State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE
SIPml-api.js?svn=222:1 setRemoteDescription(answer)
v=0
o=root 2099092377 2099092377 IN IP4 [SERVER_IP]
s=Asterisk PBX 11.16.0
c=IN IP4 [SERVER_IP]
t=0 0
m=audio 19720 RTP/SAVPF 0 8 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=ice-ufrag:028dbf70534845ca3353a97758ae64f0
a=ice-pwd:6c3e90bb5769116406a4769f51a7b134
a=candidate:Hc0a80146 1 UDP 2130706431 192.168.1.70 19720 typ host
a=candidate:S5abe665e 1 UDP 1694498815 [SERVER_IP] 60284 typ srflx raddr 192.168.1.70 rport 19720
a=candidate:Hc0a80146 2 UDP 2130706430 192.168.1.70 19721 typ host
a=candidate:S5abe665e 2 UDP 1694498814 [SERVER_IP] 53182 typ srflx raddr 192.168.1.70 rport 19721
a=connection:new
a=setup:active
a=fingerprint:SHA-256 F8:47:D2:58:1B:90:C6:91:28:3A:99:FA:1D:D5:08:2F:3E:80:09:B1:9E:6A:FC:D5:56:F5:90:AB:D9:2E:58:E6
a=sendrecv

SIPml-api.js?svn=222:1 SEND: ACK sip:1000@[SERVER_IP];transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKRnvP0X6QWmZzcm0dOcyn;rport
From: “5005"sip:5005@SERVER_IP;tag=dshEiqzXzTrTUo9auLNq
To: sip:1000@SERVER_IP;tag=as22a066e7
Contact: “5005"sip:5005@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;language=“en,fr”
Call-ID: be92af04-2b40-de63-94f6-7bba1cf8f3a5
CSeq: 18804 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username=“5005”,realm=”[SERVER_IP]”,nonce=“7e5f517e”,uri=“sip:1000@[SERVER_IP];transport=WS”,response=“22fb3eab9a5a5347263466a5058a4bad”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

SIPml-api.js?svn=222:1 ==session event = i_ao_request
SIPml-api.js?svn=222:1 onSignalingstateChange:stable
SIPml-api.js?svn=222:1 onSetRemoteDescriptionSuccess
SIPml-api.js?svn=222:1 ==session event = m_early_media
SIPml-api.js?svn=222:1 ==session event = connected
SIPml-api.js?svn=222:1 __on_add_stream
SIPml-api.js?svn=222:1 ==session event = m_stream_audio_remote_added
SIPml-api.js?svn=222:1 ==session event = m_stream_audio_local_added
SIPml-api.js?svn=222:1 ==session event = m_stream_audio_remote_added
SIPml-api.js?svn=222:1 State machine: x0000_Any_2_Trying_X_oBYE
SIPml-api.js?svn=222:1 SEND: BYE sip:1000@[SERVER_IP];transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKKKIiMiR6yuTistTe0a8tb92hbBdruwdu;rport
From: “5005"sip:5005@SERVER_IP;tag=dshEiqzXzTrTUo9auLNq
To: sip:1000@SERVER_IP;tag=as22a066e7
Call-ID: be92af04-2b40-de63-94f6-7bba1cf8f3a5
CSeq: 18805 BYE
Content-Length: 0
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language=“en,fr”
Authorization: Digest username=“5005”,realm=”[SERVER_IP]",nonce=“7e5f517e”,uri=“sip:1000@[SERVER_IP];transport=WS”,response=“b00bc6e86a1900f810fd8fbdc4161a6d”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.12.11
Organization: Doubango Telecom

SIPml-api.js?svn=222:1 PeerConnection::stop()
SIPml-api.js?svn=222:1 ==session event = terminating
SIPml-api.js?svn=222:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=222:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=54702;received=[CLIENT_IP];branch=z9hG4bKKKIiMiR6yuTistTe0a8tb92hbBdruwdu
From: "5005"sip:5005@SERVER_IP;tag=dshEiqzXzTrTUo9auLNq
To: sip:1000@SERVER_IP;tag=as22a066e7
Call-ID: be92af04-2b40-de63-94f6-7bba1cf8f3a5
CSeq: 18805 BYE
Content-Length: 0
Server: Digital-Merge_UA
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

SIPml-api.js?svn=222:1 State machine: x0000_Any_2_Terminated_X_i2xxBYE
SIPml-api.js?svn=222:1 === INVITE Dialog terminated ===
SIPml-api.js?svn=222:1 PeerConnection::stop()
SIPml-api.js?svn=222:1 ==session event = terminated
SIPml-api.js?svn=222:1 The FSM is in the final state
[/code]

RTP Debug:

Sent RTP packet to [CLIENT_IP]:54982 (type 00, seq 037333, ts 000160, len 000160) Sent RTP packet to [CLIENT_IP]:54982 (type 00, seq 037334, ts 000320, len 000160) Sent RTP packet to [CLIENT_IP]:54982 (type 00, seq 037335, ts 000480, len 000160) Sent RTP packet to [CLIENT_IP]:54982 (type 00, seq 037336, ts 000640, len 000160) Sent RTP packet to [CLIENT_IP]:54982 (type 00, seq 037337, ts 000800, len 000160) Sent RTP packet to [CLIENT_IP]:54982 (type 00, seq 037338, ts 000960, len 000160) Sent RTP packet to [CLIENT_IP]:54982 (type 00, seq 037339, ts 001120, len 000160) Sent RTP packet to [CLIENT_IP]:54982 (type 00, seq 037340, ts 001280, len 000160)

This is what wireshark is showing me:


(this “loops” as long as the session is active)

192.168.1.70 is the IP of the server behind NAT.

Any help is greatly appreciated!