Video not retransmitted: why isn't Asterisk forwarding the video data?

Hello everyone.

I’m working on an Android to Desktop (Web browser) WebRTC call implementation. I am able to successfully istantiate the call, receive and transmit audio and see the video from the desktop on the mobile. However, I am not able to see the video from the Android camera on the Desktop.

Let’s talk about this video that I am not able to see. You’ll find, attached, the Asterisk log and Wireshark data, showing that Asterisk is actually receiving the video from the Android device, but it’s not sending it to the Desktop Chrome. Using the Chrome tools, I see the only incoming data through the WebRTC connection is audio (I can see it due to low packet size).

Any idea about the cause of this issue?

[Jul 18 12:10:14] WARNING[2590]: channel.c:5522 set_format: Unable to find a codec translation path: (opus) -> (vp8|ulaw)
       > 0x7f90b814d7a0 -- Strict RTP learning after remote address set to: 37.159.184.170:63098
    -- PJSIP/624-0000003e answered PJSIP/648-0000003d
       > 0x7f90b8176780 -- Strict RTP learning after remote address set to: 37.159.184.170:50787
       > 0x7f90b8136390 -- Strict RTP learning after remote address set to: 37.159.184.170:50787
    -- Channel PJSIP/624-0000003e joined 'simple_bridge' basic-bridge <d0726308-d657-4a98-b119-5bc0971b2bb2>
    -- Channel PJSIP/648-0000003d joined 'simple_bridge' basic-bridge <d0726308-d657-4a98-b119-5bc0971b2bb2>
       > 0x7f90b814d7a0 -- Strict RTP learning after remote address set to: 192.168.5.161:58109
       > 0x7f90b814ac10 -- Strict RTP learning after ICE completion
       > 0x7f90b8176780 -- Strict RTP learning after remote address set to: 192.168.5.110:35997
       > 0x7f90b8136390 -- Strict RTP learning after ICE completion
       > 0x7f90b8136390 -- Strict RTP switching to RTP target address 192.168.5.110:35997 as source
       > 0x7f90b814ac10 -- Strict RTP switching to RTP target address 192.168.5.161:58109 as source
       > 0x7f90b814d7a0 -- Strict RTP switching to RTP target address 192.168.5.161:58109 as source
       > 0x7f90b814ac10 -- Strict RTP learning complete - Locking on source address 192.168.5.161:58109
       > 0x7f90b814d7a0 -- Strict RTP learning complete - Locking on source address 192.168.5.161:58109
       > 0x7f90b8136390 -- Strict RTP learning complete - Locking on source address 192.168.5.110:35997
    -- Channel PJSIP/648-0000003d left 'simple_bridge' basic-bridge <d0726308-d657-4a98-b119-5bc0971b2bb2>
  == Spawn extension (user-in, 624, 2) exited non-zero on 'PJSIP/648-0000003d'
    -- Executing [h@user-in:1] Hangup("PJSIP/648-0000003d", "") in new stack
  == Spawn extension (user-in, h, 1) exited non-zero on 'PJSIP/648-0000003d'
    -- Channel PJSIP/624-0000003e left 'simple_bridge' basic-bridge <d0726308-d657-4a98-b119-5bc0971b2bb2>

Your Wireshark is filtered to only show one flow. Have you also looked at “rtp set debug on” to see if it is actually being received and decoded? Have you done a test not in a call with the device having a problem using Echo to echo back received media?

You basically have to isolate it down.

Also didn’t you create a post a week ago with this same problem and I responded with the same last thing?

Hello jcolp,
thanks a lot for helping here.
I also tried calling the Echo and the scenario is exactly the same.
Practically, I see on asterisk:

== Setting global variable ‘SIPDOMAIN’ to ‘videoconf.i-tel.it’
== DTLS ECDH initialized (automatic), faster PFS enabled
– Executing [echo@user-in:1] Answer(“PJSIP/648-00000052”, “”) in new stack
> 0x7f90b816dad0 – Strict RTP learning after remote address set to: 192.168.5.110:59071
> 0x7f90b8136390 – Strict RTP learning after remote address set to: 192.168.5.110:59071
> 0x7f90b816dad0 – Strict RTP learning after remote address set to: 192.168.5.110:59071
> 0x7f90b8136390 – Strict RTP learning after ICE completion
– Executing [echo@user-in:2] Echo(“PJSIP/648-00000052”, “”) in new stack
> 0x7f90b8136390 – Strict RTP switching to RTP target address 192.168.5.110:59071 as source
> 0x7f90b8136390 – Strict RTP learning complete - Locking on source address 192.168.5.110:59071

and only audio is echoed back to android.
I don’t have the video stream despite snoop shows this is sent from android to asterisk on same port as audio (59071).
What I also noticed is that asterisk is only locking on one source address while I would have expected one for audio and one for video
on same link.
No error messages appearing on asterisk to clearly identifying an issue.
I have also attached snoop where .110 is android and .220 is asterisk.
What could be the problem?

Thanks again

What is actually negotiated? What is the SIP trace?

I had another run of calling Echo this time with PJSIP logger enabled. Hope this is helpful. Thanks again.


<--- Received SIP request (4772 bytes) from WSS:192.168.5.110:47833 --->
INVITE sip:echo@videoconf.i-tel.it SIP/2.0
Via: SIP/2.0/WSS videoconf.i-tel.it;branch=z9hG4bK5327985
Max-Forwards: 69
To: <sip:echo@videoconf.i-tel.it>
From: <sip:648@videoconf.i-tel.it>;tag=lhnd245ih8
Call-ID: k7k77efjc6sqlhfo4s37
CSeq: 8864 INVITE
Contact: <sip:648@videoconf.i-tel.it;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.2.10
Content-Length: 4267

v=0
o=- 3908805219845887234 4 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS f0baaea8-9b50-46ff-b2d7-d05ced1a3248
m=audio 47532 UDP/TLS/RTP/SAVPF 111 103 9 102 0 8 105 13 110 113 126
c=IN IP4 192.168.5.110
a=rtcp:47861 IN IP4 192.168.5.110
a=candidate:177172242 1 udp 2122260223 192.168.5.110 47532 typ host generation 0 network-id 5 network-cost 10
a=candidate:177172242 1 udp 2122194687 192.168.5.110 48901 typ host generation 0 network-id 3 network-cost 900
a=candidate:177172242 2 udp 2122260222 192.168.5.110 47861 typ host generation 0 network-id 5 network-cost 10
a=candidate:177172242 2 udp 2122194686 192.168.5.110 48149 typ host generation 0 network-id 3 network-cost 900
a=ice-ufrag:uc6c
a=ice-pwd:Ymlet331wUU5AHKIqPSJ4BGn
a=ice-options:trickle renomination
a=fingerprint:sha-256 51:DF:A6:9A:16:25:4E:2E:A5:8A:43:F3:B4:74:51:9D:22:7D:35:1B:41:3C:4D:05:F6:21:F9:C9:8B:CA:14:72
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2109724827 cname:rroU2pPZWFaYfOsC
a=ssrc:2109724827 msid:f0baaea8-9b50-46ff-b2d7-d05ced1a3248 53690e5c-d145-439c-82a1-d0778b9909a6
a=ssrc:2109724827 mslabel:f0baaea8-9b50-46ff-b2d7-d05ced1a3248
a=ssrc:2109724827 label:53690e5c-d145-439c-82a1-d0778b9909a6
m=video 56777 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 127 125 104
c=IN IP4 192.168.5.110
a=rtcp:41072 IN IP4 192.168.5.110
a=candidate:177172242 1 udp 2122260223 192.168.5.110 56777 typ host generation 0 network-id 5 network-cost 10
a=candidate:177172242 1 udp 2122194687 192.168.5.110 37847 typ host generation 0 network-id 3 network-cost 900
a=candidate:177172242 2 udp 2122260222 192.168.5.110 41072 typ host generation 0 network-id 5 network-cost 10
a=candidate:177172242 2 udp 2122194686 192.168.5.110 38949 typ host generation 0 network-id 3 network-cost 900
a=ice-ufrag:uc6c
a=ice-pwd:Ymlet331wUU5AHKIqPSJ4BGn
a=ice-options:trickle renomination
a=fingerprint:sha-256 51:DF:A6:9A:16:25:4E:2E:A5:8A:43:F3:B4:74:51:9D:22:7D:35:1B:41:3C:4D:05:F6:21:F9:C9:8B:CA:14:72
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 red/90000
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:127 ulpfec/90000
a=rtpmap:125 H264/90000
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:104 rtx/90000
a=fmtp:104 apt=125
a=ssrc-group:FID 3579692342 448016958
a=ssrc:3579692342 cname:rroU2pPZWFaYfOsC
a=ssrc:3579692342 msid:f0baaea8-9b50-46ff-b2d7-d05ced1a3248 ac6340e1-044b-4c80-b76c-75d6945236e7
a=ssrc:3579692342 mslabel:f0baaea8-9b50-46ff-b2d7-d05ced1a3248
a=ssrc:3579692342 label:ac6340e1-044b-4c80-b76c-75d6945236e7
a=ssrc:448016958 cname:rroU2pPZWFaYfOsC
a=ssrc:448016958 msid:f0baaea8-9b50-46ff-b2d7-d05ced1a3248 ac6340e1-044b-4c80-b76c-75d6945236e7
a=ssrc:448016958 mslabel:f0baaea8-9b50-46ff-b2d7-d05ced1a3248
a=ssrc:448016958 label:ac6340e1-044b-4c80-b76c-75d6945236e7

<--- Transmitting SIP response (473 bytes) to WSS:192.168.5.110:47833 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS videoconf.i-tel.it;rport=47833;received=192.168.5.110;branch=z9hG4bK5327985
Call-ID: k7k77efjc6sqlhfo4s37
From: <sip:648@videoconf.i-tel.it>;tag=lhnd245ih8
To: <sip:echo@videoconf.i-tel.it>;tag=z9hG4bK5327985
CSeq: 8864 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1531917663/834d8a14838800aff0b3d62b20bf1c0c",opaque="6d207e6243a75bb1",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (402 bytes) from WSS:192.168.5.110:47833 --->
ACK sip:echo@videoconf.i-tel.it SIP/2.0
Via: SIP/2.0/WSS videoconf.i-tel.it;branch=z9hG4bK5327985
Max-Forwards: 69
To: <sip:echo@videoconf.i-tel.it>;tag=z9hG4bK5327985
From: <sip:648@videoconf.i-tel.it>;tag=lhnd245ih8
Call-ID: k7k77efjc6sqlhfo4s37
CSeq: 8864 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.2.10
Content-Length: 0


<--- Received SIP request (5049 bytes) from WSS:192.168.5.110:47833 --->
INVITE sip:echo@videoconf.i-tel.it SIP/2.0
Via: SIP/2.0/WSS videoconf.i-tel.it;branch=z9hG4bK9654333
Max-Forwards: 69
To: <sip:echo@videoconf.i-tel.it>
From: <sip:648@videoconf.i-tel.it>;tag=lhnd245ih8
Call-ID: k7k77efjc6sqlhfo4s37
CSeq: 8865 INVITE
Authorization: Digest algorithm=MD5, username="648", realm="asterisk", nonce="1531917663/834d8a14838800aff0b3d62b20bf1c0c", uri="sip:echo@videoconf.i-tel.it", response="e5b40c58131b0f2806e191b26932e047", opaque="6d207e6243a75bb1", qop=auth, cnonce="hbe7ca3rt6c2", nc=00000001
Contact: <sip:648@videoconf.i-tel.it;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.2.10
Content-Length: 4267

v=0
o=- 3908805219845887234 4 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS f0baaea8-9b50-46ff-b2d7-d05ced1a3248
m=audio 47532 UDP/TLS/RTP/SAVPF 111 103 9 102 0 8 105 13 110 113 126
c=IN IP4 192.168.5.110
a=rtcp:47861 IN IP4 192.168.5.110
a=candidate:177172242 1 udp 2122260223 192.168.5.110 47532 typ host generation 0 network-id 5 network-cost 10
a=candidate:177172242 1 udp 2122194687 192.168.5.110 48901 typ host generation 0 network-id 3 network-cost 900
a=candidate:177172242 2 udp 2122260222 192.168.5.110 47861 typ host generation 0 network-id 5 network-cost 10
a=candidate:177172242 2 udp 2122194686 192.168.5.110 48149 typ host generation 0 network-id 3 network-cost 900
a=ice-ufrag:uc6c
a=ice-pwd:Ymlet331wUU5AHKIqPSJ4BGn
a=ice-options:trickle renomination
a=fingerprint:sha-256 51:DF:A6:9A:16:25:4E:2E:A5:8A:43:F3:B4:74:51:9D:22:7D:35:1B:41:3C:4D:05:F6:21:F9:C9:8B:CA:14:72
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:102 ILBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2109724827 cname:rroU2pPZWFaYfOsC
a=ssrc:2109724827 msid:f0baaea8-9b50-46ff-b2d7-d05ced1a3248 53690e5c-d145-439c-82a1-d0778b9909a6
a=ssrc:2109724827 mslabel:f0baaea8-9b50-46ff-b2d7-d05ced1a3248
a=ssrc:2109724827 label:53690e5c-d145-439c-82a1-d0778b9909a6
m=video 56777 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 127 125 104
c=IN IP4 192.168.5.110
a=rtcp:41072 IN IP4 192.168.5.110
a=candidate:177172242 1 udp 2122260223 192.168.5.110 56777 typ host generation 0 network-id 5 network-cost 10
a=candidate:177172242 1 udp 2122194687 192.168.5.110 37847 typ host generation 0 network-id 3 network-cost 900
a=candidate:177172242 2 udp 2122260222 192.168.5.110 41072 typ host generation 0 network-id 5 network-cost 10
a=candidate:177172242 2 udp 2122194686 192.168.5.110 38949 typ host generation 0 network-id 3 network-cost 900
a=ice-ufrag:uc6c
a=ice-pwd:Ymlet331wUU5AHKIqPSJ4BGn
a=ice-options:trickle renomination
a=fingerprint:sha-256 51:DF:A6:9A:16:25:4E:2E:A5:8A:43:F3:B4:74:51:9D:22:7D:35:1B:41:3C:4D:05:F6:21:F9:C9:8B:CA:14:72
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=sendrecv
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 red/90000
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:127 ulpfec/90000
a=rtpmap:125 H264/90000
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=rtcp-fb:125 goog-remb
a=rtcp-fb:125 transport-cc
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:104 rtx/90000
a=fmtp:104 apt=125
a=ssrc-group:FID 3579692342 448016958
a=ssrc:3579692342 cname:rroU2pPZWFaYfOsC
a=ssrc:3579692342 msid:f0baaea8-9b50-46ff-b2d7-d05ced1a3248 ac6340e1-044b-4c80-b76c-75d6945236e7
a=ssrc:3579692342 mslabel:f0baaea8-9b50-46ff-b2d7-d05ced1a3248
a=ssrc:3579692342 label:ac6340e1-044b-4c80-b76c-75d6945236e7
a=ssrc:448016958 cname:rroU2pPZWFaYfOsC
a=ssrc:448016958 msid:f0baaea8-9b50-46ff-b2d7-d05ced1a3248 ac6340e1-044b-4c80-b76c-75d6945236e7
a=ssrc:448016958 mslabel:f0baaea8-9b50-46ff-b2d7-d05ced1a3248
a=ssrc:448016958 label:ac6340e1-044b-4c80-b76c-75d6945236e7

  == Setting global variable 'SIPDOMAIN' to 'videoconf.i-tel.it'
<--- Transmitting SIP response (301 bytes) to WSS:192.168.5.110:47833 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS videoconf.i-tel.it;rport=47833;received=192.168.5.110;branch=z9hG4bK9654333
Call-ID: k7k77efjc6sqlhfo4s37
From: <sip:648@videoconf.i-tel.it>;tag=lhnd245ih8
To: <sip:echo@videoconf.i-tel.it>
CSeq: 8865 INVITE
Server: Asterisk PBX 15.3.0
Content-Length:  0


  == DTLS ECDH initialized (automatic), faster PFS enabled
    -- Executing [echo@user-in:1] Answer("PJSIP/648-00000055", "") in new stack
       > 0x7f90c0072d70 -- Strict RTP learning after remote address set to: 192.168.5.110:47532
       > 0x7f90c006fa00 -- Strict RTP learning after remote address set to: 192.168.5.110:47532
<--- Transmitting SIP response (2005 bytes) to WSS:192.168.5.110:47833 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS videoconf.i-tel.it;rport=47833;received=192.168.5.110;branch=z9hG4bK9654333
Call-ID: k7k77efjc6sqlhfo4s37
From: <sip:648@videoconf.i-tel.it>;tag=lhnd245ih8
To: <sip:echo@videoconf.i-tel.it>;tag=3f0fd89b-1b15-4900-b317-4dff948db4fd
CSeq: 8865 INVITE
Server: Asterisk PBX 15.3.0
Contact: <sip:192.168.5.220:8443;transport=ws>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:  1375

v=0
o=- 1092310274 6 IN IP4 192.168.5.220
s=Asterisk
c=IN IP4 192.168.5.220
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio video
m=audio 54356 UDP/TLS/RTP/SAVPF 0 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 D2:DE:DC:2B:8E:1E:85:C9:89:DB:1C:2D:7D:8F:C0:72:A6:C1:33:75:01:E1:0C:5F:90:05:37:7E:AE:2C:21:4C
a=ice-ufrag:012ffb4869c00a233d1495c2601a7197
a=ice-pwd:382e258f2269a9836f280d712f7280b3
a=candidate:Hc0a805dc 1 UDP 2130706431 192.168.5.220 54356 typ host
a=candidate:R259fb8aa 1 UDP 16777215 37.159.184.170 56441 typ relay raddr 192.168.5.220 rport 59061
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:1687697238 cname:14bcda1e-1e0b-432c-9e92-801e4efc7d7d
a=msid:831bba81-23ca-4783-ab3a-444d6536743a 69b2b6fd-d974-471c-91b6-d714d3ed6dff
a=mid:audio
m=video 54356 UDP/TLS/RTP/SAVPF 96
a=connection:new
a=setup:active
a=fingerprint:SHA-256 D2:DE:DC:2B:8E:1E:85:C9:89:DB:1C:2D:7D:8F:C0:72:A6:C1:33:75:01:E1:0C:5F:90:05:37:7E:AE:2C:21:4C
a=ice-ufrag:012ffb4869c00a233d1495c2601a7197
a=ice-pwd:382e258f2269a9836f280d712f7280b3
a=rtpmap:96 VP8/90000
a=sendrecv
a=rtcp-mux
a=ssrc:349547932 cname:943eb275-3bae-4715-9e45-d5283cb05b6c
a=msid:831bba81-23ca-4783-ab3a-444d6536743a 0fa9b64a-4568-4900-a041-6b53ec903a7d
a=rtcp-fb:* ccm fir
a=mid:video

       > 0x7f90c0072d70 -- Strict RTP learning after remote address set to: 192.168.5.110:47532
       > 0x7f90c006fa00 -- Strict RTP learning after ICE completion
<--- Transmitting SIP response (2005 bytes) to WSS:192.168.5.110:47833 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS videoconf.i-tel.it;rport=47833;received=192.168.5.110;branch=z9hG4bK9654333
Call-ID: k7k77efjc6sqlhfo4s37
From: <sip:648@videoconf.i-tel.it>;tag=lhnd245ih8
To: <sip:echo@videoconf.i-tel.it>;tag=3f0fd89b-1b15-4900-b317-4dff948db4fd
CSeq: 8865 INVITE
Server: Asterisk PBX 15.3.0
Contact: <sip:192.168.5.220:8443;transport=ws>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:  1375

v=0
o=- 1092310274 6 IN IP4 192.168.5.220
s=Asterisk
c=IN IP4 192.168.5.220
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio video
m=audio 54356 UDP/TLS/RTP/SAVPF 0 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 D2:DE:DC:2B:8E:1E:85:C9:89:DB:1C:2D:7D:8F:C0:72:A6:C1:33:75:01:E1:0C:5F:90:05:37:7E:AE:2C:21:4C
a=ice-ufrag:012ffb4869c00a233d1495c2601a7197
a=ice-pwd:382e258f2269a9836f280d712f7280b3
a=candidate:Hc0a805dc 1 UDP 2130706431 192.168.5.220 54356 typ host
a=candidate:R259fb8aa 1 UDP 16777215 37.159.184.170 56441 typ relay raddr 192.168.5.220 rport 59061
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
a=ssrc:1687697238 cname:14bcda1e-1e0b-432c-9e92-801e4efc7d7d
a=msid:831bba81-23ca-4783-ab3a-444d6536743a 69b2b6fd-d974-471c-91b6-d714d3ed6dff
a=mid:audio
m=video 54356 UDP/TLS/RTP/SAVPF 96
a=connection:new
a=setup:active
a=fingerprint:SHA-256 D2:DE:DC:2B:8E:1E:85:C9:89:DB:1C:2D:7D:8F:C0:72:A6:C1:33:75:01:E1:0C:5F:90:05:37:7E:AE:2C:21:4C
a=ice-ufrag:012ffb4869c00a233d1495c2601a7197
a=ice-pwd:382e258f2269a9836f280d712f7280b3
a=rtpmap:96 VP8/90000
a=sendrecv
a=rtcp-mux
a=ssrc:349547932 cname:943eb275-3bae-4715-9e45-d5283cb05b6c
a=msid:831bba81-23ca-4783-ab3a-444d6536743a 0fa9b64a-4568-4900-a041-6b53ec903a7d
a=rtcp-fb:* ccm fir
a=mid:video

    -- Executing [echo@user-in:2] Echo("PJSIP/648-00000055", "") in new stack
       > 0x7f90c006fa00 -- Strict RTP switching to RTP target address 192.168.5.110:47532 as source
<--- Received SIP request (431 bytes) from WSS:192.168.5.110:47833 --->
ACK sip:192.168.5.220:8443;transport=ws SIP/2.0
Via: SIP/2.0/WSS videoconf.i-tel.it;branch=z9hG4bK574702
Max-Forwards: 69
To: <sip:echo@videoconf.i-tel.it>;tag=3f0fd89b-1b15-4900-b317-4dff948db4fd
From: <sip:648@videoconf.i-tel.it>;tag=lhnd245ih8
Call-ID: k7k77efjc6sqlhfo4s37
CSeq: 8865 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.2.10
Content-Length: 0


<--- Received SIP request (432 bytes) from WSS:192.168.5.110:47833 --->
ACK sip:192.168.5.220:8443;transport=ws SIP/2.0
Via: SIP/2.0/WSS videoconf.i-tel.it;branch=z9hG4bK8426945
Max-Forwards: 69
To: <sip:echo@videoconf.i-tel.it>;tag=3f0fd89b-1b15-4900-b317-4dff948db4fd
From: <sip:648@videoconf.i-tel.it>;tag=lhnd245ih8
Call-ID: k7k77efjc6sqlhfo4s37
CSeq: 8865 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.2.10
Content-Length: 0


       > 0x7f90c006fa00 -- Strict RTP learning complete - Locking on source address 192.168.5.110:47532
<--- Received SIP request (532 bytes) from UDP:192.168.5.40:5060 --->
OPTIONS sip:192.168.5.220 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.40:5060;branch=z9hG4bK4feed4f4
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.5.40>;tag=as19db39c7
To: <sip:192.168.5.220>
Contact: <sip:asterisk@192.168.5.40:5060>
Call-ID: 20a8261b2e2c48741a36f6ca2e804067@192.168.5.40:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.8.2
Date: Wed, 18 Jul 2018 12:33:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<--- Transmitting SIP response (832 bytes) to UDP:192.168.5.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.40:5060;rport=5060;received=192.168.5.40;branch=z9hG4bK4feed4f4
Call-ID: 20a8261b2e2c48741a36f6ca2e804067@192.168.5.40:5060
From: "asterisk" <sip:asterisk@192.168.5.40>;tag=as19db39c7
To: <sip:192.168.5.220>;tag=z9hG4bK4feed4f4
CSeq: 102 OPTIONS
Accept: application/sdp, application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/pidf+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, message/sipfrag;version=2.0
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: text/plain
Accept-Language: en
Server: Asterisk PBX 15.3.0
Content-Length:  0


<--- Received SIP request (432 bytes) from WSS:192.168.5.110:47833 --->
BYE sip:192.168.5.220:8443;transport=ws SIP/2.0
Via: SIP/2.0/WSS videoconf.i-tel.it;branch=z9hG4bK1461096
Max-Forwards: 69
To: <sip:echo@videoconf.i-tel.it>;tag=3f0fd89b-1b15-4900-b317-4dff948db4fd
From: <sip:648@videoconf.i-tel.it>;tag=lhnd245ih8
Call-ID: k7k77efjc6sqlhfo4s37
CSeq: 8866 BYE
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
Supported: outbound
User-Agent: JsSIP 3.2.10
Content-Length: 0


<--- Transmitting SIP response (335 bytes) to WSS:192.168.5.110:47833 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS videoconf.i-tel.it;rport=47833;received=192.168.5.110;branch=z9hG4bK1461096
Call-ID: k7k77efjc6sqlhfo4s37
From: <sip:648@videoconf.i-tel.it>;tag=lhnd245ih8
To: <sip:echo@videoconf.i-tel.it>;tag=3f0fd89b-1b15-4900-b317-4dff948db4fd
CSeq: 8866 BYE
Server: Asterisk PBX 15.3.0
Content-Length:  0


  == Spawn extension (user-in, echo, 2) exited non-zero on 'PJSIP/648-00000055'
    -- Executing [h@user-in:1] Hangup("PJSIP/648-00000055", "") in new stack
  == Spawn extension (user-in, h, 1) exited non-zero on 'PJSIP/648-00000055'

Nothing stands out. We do this every day, so it’s likely something browser side or specific to your configuration. You’d need to dig into the code most likely.

Hello,
I’m stumbling on the same issue and I can confirm that with FF this works perfectly.
It seems a specific problem between Chrome and Asterisk but can’t figure out which of the two is to blame.
I only noticed that for the video part, Chrome adds the ssrc-group:FID attribute which is missing in the FF’s SDP.
Does Asterisk/pjsip support that?
Any other troubleshooting I could run to find rootcause?

Thanks