Webrtc SipML 5 audio issue + asterisk

Asterisk Asterisk WebRTC
1

Hi,
We are using asterisk 16.0.7 + webrtc for video calling but facing the issue of no audio in case when caller dial to callee… in this case we ain’t get any audio at callee side.
if caller (audio YES/ video YES)-------> callee.(video YES , audio NO)

can any one suggest were we are doing wrong
below is our configuration :

pjsip.conf

[webrtc_client]
type=aor
max_contacts=1
remove_existing=yes

[webrtc_client]
type=auth
auth_type=userpass
username=webrtc_client
password=walk

[webrtc_client]
type=endpoint
aors=webrtc_client
;transport=transport-wss
auth=webrtc_client
;dtls_auto_generate_cert=yes
webrtc=yes
from_domain= sip.xxxxx.com
context=internal
disallow=all
allow=opus,vp9,ulaw,g722,vp8,h264
;allow=g722,ulaw,vp9,vp8,h264
;allow=g722,vp9
force_rport=yes
direct_media=no
rtp_symmetric=yes
max_audio_streams=10
max_video_streams=10
;ice_support=yes
;rewrite_contact=yes
2

Check the negotiations (SIP trace and SDP), and make sure things negotiated as expected. Then enable “rtp debug” in Asterisk and check to make sure the audio is going to/from the expected address and ports.

3

Hi, thanks @kharwell, following is the sip trace for a call from webrtc_client02 to webrtc_client03 both of which were in the same network while the asterisk server is hosted on cloud, I took the one from pjsip logger cause there were some errors which I thought might be useful.

<--- Received SIP request (5794 bytes) from WS:122.175.175.96:43212 --->
INVITE sip:webrtc_client03@sip.mysipdomain.com SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKs2oGASkcV3GA72AvAmQP1F4EHWxudxNM;rport
From: <sip:webrtc_client02@sip.mysipdomain.com>;tag=O8nAOtKBaGyB9nDHVB9J
To: <sip:webrtc_client03@sip.mysipdomain.com>
Contact: <sips:webrtc_client02@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 39997fb9-bff0-4fdc-dd5b-5a1fbfa1f0bd
CSeq: 17058 INVITE
Content-Type: application/sdp
Content-Length: 5183
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=- 4460269225836751400 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE 0 1
a=msid-semantic: WMS D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI
m=audio 44100 UDP/TLS/RTP/SAVPF 111 103 9 0 8 105 13 110 113 126
c=IN IP4 192.168.1.4
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3022624816 1 udp 2122260223 192.168.1.4 44100 typ host generation 0 network-id 2 network-cost 10
a=candidate:3068586373 1 udp 2122197247 2401:4900:1ba8:cf5:cad3:483f:23a2:4263 45799 typ host generation 0 network-id 1 network-cost 900
a=candidate:4205470912 1 tcp 1518280447 192.168.1.4 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:4167374197 1 tcp 1518217471 2401:4900:1ba8:cf5:cad3:483f:23a2:4263 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:Fgnm
a=ice-pwd:95v++eWwML3NsWX12K2twQB+
a=ice-options:trickle
a=fingerprint:sha-256 50:FB:B9:2A:6B:C9:1F:48:B9:13:C6:78:33:ED:D6:19:D3:1C:CA:ED:F8:72:4D:3F:A0:E6:CC:66:9E:36:88:34
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI 66684f82-9aed-4945-957e-8b5288400e85
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: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:4029574070 cname:cweiZIYnVH8l0eUn
a=ssrc:4029574070 msid:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI 66684f82-9aed-4945-957e-8b5288400e85
a=ssrc:4029574070 mslabel:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI
a=ssrc:4029574070 label:66684f82-9aed-4945-957e-8b5288400e85
m=video 42217 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 123 104
c=IN IP4 192.168.1.4
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3022624816 1 udp 2122260223 192.168.1.4 42217 typ host generation 0 network-id 2 network-cost 10
a=candidate:3068586373 1 udp 2122197247 2401:4900:1ba8:cf5:cad3:483f:23a2:4263 42497 typ host generation 0 network-id 1 network-cost 900
a=candidate:4205470912 1 tcp 1518280447 192.168.1.4 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:4167374197 1 tcp 1518217471 2401:4900:1ba8:cf5:cad3:483f:23a2:4263 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:Fgnm
a=ice-pwd:95v++eWwML3NsWX12K2twQB+
a=ice-options:trickle
a=fingerprint:sha-256 50:FB:B9:2A:6B:C9:1F:48:B9:13:C6:78:33:ED:D6:19:D3:1C:CA:ED:F8:72:4D:3F:A0:E6:CC:66:9E:36:88:34
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI b1f8614a-fe6c-4a9c-ac2c-f75326431086
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 H264/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 red/90000
a=rtpmap:123 rtx/90000
a=fmtp:123 apt=102
a=rtpmap:104 ulpfec/90000
a=ssrc-group:FID 1754551035 1833096269
a=ssrc:1754551035 cname:cweiZIYnVH8l0eUn
a=ssrc:1754551035 msid:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI b1f8614a-fe6c-4a9c-ac2c-f75326431086
a=ssrc:1754551035 mslabel:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI
a=ssrc:1754551035 label:b1f8614a-fe6c-4a9c-ac2c-f75326431086
a=ssrc:1833096269 cname:cweiZIYnVH8l0eUn
a=ssrc:1833096269 msid:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI b1f8614a-fe6c-4a9c-ac2c-f75326431086
a=ssrc:1833096269 mslabel:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI
a=ssrc:1833096269 label:b1f8614a-fe6c-4a9c-ac2c-f75326431086

<--- Transmitting SIP response (569 bytes) to WS:122.175.175.96:43212 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=43212;received=122.175.175.96;branch=z9hG4bKs2oGASkcV3GA72AvAmQP1F4EHWxudxNM
Call-ID: 39997fb9-bff0-4fdc-dd5b-5a1fbfa1f0bd
From: <sip:webrtc_client02@sip.mysipdomain.com>;tag=O8nAOtKBaGyB9nDHVB9J
To: <sip:webrtc_client03@sip.mysipdomain.com>;tag=z9hG4bKs2oGASkcV3GA72AvAmQP1F4EHWxudxNM
CSeq: 17058 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1587676512/937cab544e0ef68c9279cdf90d7a9913",opaque="51aac44b1493c5eb",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.4.1
Content-Length:  0


<--- Received SIP request (401 bytes) from WS:122.175.175.96:43212 --->
ACK sip:webrtc_client03@sip.mysipdomain.com SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKs2oGASkcV3GA72AvAmQP1F4EHWxudxNM;rport
From: <sip:webrtc_client02@sip.mysipdomain.com>;tag=O8nAOtKBaGyB9nDHVB9J
To: <sip:webrtc_client03@sip.mysipdomain.com>;tag=z9hG4bKs2oGASkcV3GA72AvAmQP1F4EHWxudxNM
Call-ID: 39997fb9-bff0-4fdc-dd5b-5a1fbfa1f0bd
CSeq: 17058 ACK
Content-Length: 0
Max-Forwards: 70


<--- Received SIP request (6102 bytes) from WS:122.175.175.96:43212 --->
INVITE sip:webrtc_client03@sip.mysipdomain.com SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKEmIIJTrJUqxwP7QR4kQR0Mq0NUSf7TdF;rport
From: <sip:webrtc_client02@sip.mysipdomain.com>;tag=O8nAOtKBaGyB9nDHVB9J
To: <sip:webrtc_client03@sip.mysipdomain.com>
Contact: <sips:webrtc_client02@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 39997fb9-bff0-4fdc-dd5b-5a1fbfa1f0bd
CSeq: 17059 INVITE
Content-Type: application/sdp
Content-Length: 5183
Max-Forwards: 70
Authorization: Digest username="webrtc_client02",realm="asterisk",nonce="1587676512/937cab544e0ef68c9279cdf90d7a9913",uri="sip:webrtc_client03@sip.mysipdomain.com",response="76130774de75dfa5265cd09bd8429030",algorithm=md5,cnonce="da32e7ef04ef4a36eebeb333f35deece",opaque="51aac44b1493c5eb",qop=auth,nc=00000001
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=- 4460269225836751400 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE 0 1
a=msid-semantic: WMS D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI
m=audio 44100 UDP/TLS/RTP/SAVPF 111 103 9 0 8 105 13 110 113 126
c=IN IP4 192.168.1.4
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3022624816 1 udp 2122260223 192.168.1.4 44100 typ host generation 0 network-id 2 network-cost 10
a=candidate:3068586373 1 udp 2122197247 2401:4900:1ba8:cf5:cad3:483f:23a2:4263 45799 typ host generation 0 network-id 1 network-cost 900
a=candidate:4205470912 1 tcp 1518280447 192.168.1.4 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:4167374197 1 tcp 1518217471 2401:4900:1ba8:cf5:cad3:483f:23a2:4263 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:Fgnm
a=ice-pwd:95v++eWwML3NsWX12K2twQB+
a=ice-options:trickle
a=fingerprint:sha-256 50:FB:B9:2A:6B:C9:1F:48:B9:13:C6:78:33:ED:D6:19:D3:1C:CA:ED:F8:72:4D:3F:A0:E6:CC:66:9E:36:88:34
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI 66684f82-9aed-4945-957e-8b5288400e85
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: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:4029574070 cname:cweiZIYnVH8l0eUn
a=ssrc:4029574070 msid:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI 66684f82-9aed-4945-957e-8b5288400e85
a=ssrc:4029574070 mslabel:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI
a=ssrc:4029574070 label:66684f82-9aed-4945-957e-8b5288400e85
m=video 42217 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 123 104
c=IN IP4 192.168.1.4
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3022624816 1 udp 2122260223 192.168.1.4 42217 typ host generation 0 network-id 2 network-cost 10
a=candidate:3068586373 1 udp 2122197247 2401:4900:1ba8:cf5:cad3:483f:23a2:4263 42497 typ host generation 0 network-id 1 network-cost 900
a=candidate:4205470912 1 tcp 1518280447 192.168.1.4 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:4167374197 1 tcp 1518217471 2401:4900:1ba8:cf5:cad3:483f:23a2:4263 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:Fgnm
a=ice-pwd:95v++eWwML3NsWX12K2twQB+
a=ice-options:trickle
a=fingerprint:sha-256 50:FB:B9:2A:6B:C9:1F:48:B9:13:C6:78:33:ED:D6:19:D3:1C:CA:ED:F8:72:4D:3F:A0:E6:CC:66:9E:36:88:34
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI b1f8614a-fe6c-4a9c-ac2c-f75326431086
a=rtcp-mux
a=rtcp-rsize
a=rtpmap:96 VP8/90000
a=rtcp-fb:96 goog-remb
a=rtcp-fb:96 transport-cc
a=rtcp-fb:96 ccm fir
a=rtcp-fb:96 nack
a=rtcp-fb:96 nack pli
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96
a=rtpmap:98 VP9/90000
a=rtcp-fb:98 goog-remb
a=rtcp-fb:98 transport-cc
a=rtcp-fb:98 ccm fir
a=rtcp-fb:98 nack
a=rtcp-fb:98 nack pli
a=fmtp:98 profile-id=0
a=rtpmap:99 rtx/90000
a=fmtp:99 apt=98
a=rtpmap:100 H264/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:101 rtx/90000
a=fmtp:101 apt=100
a=rtpmap:102 red/90000
a=rtpmap:123 rtx/90000
a=fmtp:123 apt=102
a=rtpmap:104 ulpfec/90000
a=ssrc-group:FID 1754551035 1833096269
a=ssrc:1754551035 cname:cweiZIYnVH8l0eUn
a=ssrc:1754551035 msid:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI b1f8614a-fe6c-4a9c-ac2c-f75326431086
a=ssrc:1754551035 mslabel:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI
a=ssrc:1754551035 label:b1f8614a-fe6c-4a9c-ac2c-f75326431086
a=ssrc:1833096269 cname:cweiZIYnVH8l0eUn
a=ssrc:1833096269 msid:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI b1f8614a-fe6c-4a9c-ac2c-f75326431086
a=ssrc:1833096269 mslabel:D0PWso95zlBc1Yr8dGSrLKecB7BFGMkpBzkI
a=ssrc:1833096269 label:b1f8614a-fe6c-4a9c-ac2c-f75326431086

  == Setting global variable 'SIPDOMAIN' to 'sip.mysipdomain.com'
<--- Transmitting SIP response (373 bytes) to WS:122.175.175.96:43212 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=43212;received=122.175.175.96;branch=z9hG4bKEmIIJTrJUqxwP7QR4kQR0Mq0NUSf7TdF
Call-ID: 39997fb9-bff0-4fdc-dd5b-5a1fbfa1f0bd
From: <sip:webrtc_client02@sip.mysipdomain.com>;tag=O8nAOtKBaGyB9nDHVB9J
To: <sip:webrtc_client03@sip.mysipdomain.com>
CSeq: 17059 INVITE
Server: Asterisk PBX 16.4.1
Content-Length:  0


    -- Executing [webrtc_client03@internal:1] Log("PJSIP/webrtc_client02-000000c1", "NOTICE, Dialing from "" <webrtc_client02>) to webrtc_client03") in new stack
[2020-04-23 21:15:13] NOTICE[13379][C-00000067]: Ext. webrtc_client03:1 @ internal:  Dialing from "" <webrtc_client02>) to webrtc_client03
    -- Executing [webrtc_client03@internal:2] Dial("PJSIP/webrtc_client02-000000c1", "PJSIP/webrtc_client03") in new stack
    -- Called PJSIP/webrtc_client03
<--- Transmitting SIP request (2400 bytes) to WS:122.175.175.96:37529 --->
INVITE sips:webrtc_client03@122.175.175.96:37529;transport=ws;rtcweb-breaker=no SIP/2.0
Via: SIP/2.0/WS 10.0.11.55:8088;rport;branch=z9hG4bKPj1173b431-c1ca-49d5-93fc-6fb09d95e266;alias
From: <sip:webrtc_client02@ip-10-0-11-55>;tag=4304a09f-efb4-4089-98bb-d270e4341fe7
To: <sips:webrtc_client03@122.175.175.96;rtcweb-breaker=no>
Contact: <sips:asterisk@ip-10-0-11-55:5060;transport=ws>
Call-ID: 1fdf6f6b-b778-4026-b7fc-b94f8f35e11f
CSeq: 2367 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.4.1
Content-Type: application/sdp
Content-Length:  1643

v=0
o=- 2021835922 2021835922 IN IP4 10.0.11.55
s=Asterisk
c=IN IP4 10.0.11.55
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio-0 video-1
m=audio 11184 UDP/TLS/RTP/SAVPF 0 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 BE:E3:74:CF:40:7B:87:6D:68:99:0E:1C:FB:84:7A:F9:D6:36:10:6F:42:82:55:4B:43:20:66:D1:64:CF:D0:76
a=ice-ufrag:2ad4320669f8831c24d2b0446ff212a4
a=ice-pwd:188bb68d2c0238fa3b95d9d65a5fa572
a=candidate:H821e2c3e 1 UDP 2130706431 fe80::8f3:f5ff:fee0:bd36 11184 typ host
a=candidate:Ha000b37 1 UDP 2130706431 10.0.11.55 11184 typ host
a=candidate:S22f4393b 1 UDP 1694498815 34.244.57.59 11184 typ srflx raddr 10.0.11.55 rport 11184
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:484045528 cname:f2f43940-dfed-44ba-976c-fe5d6800efea
a=msid:5301ff2b-a48a-4c44-a0e0-c33a19bf3ecd 9441576d-53c0-4de4-844b-39539dab8401
a=rtcp-fb:* transport-cc
a=mid:audio-0
m=video 11184 UDP/TLS/RTP/SAVPF 100
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 BE:E3:74:CF:40:7B:87:6D:68:99:0E:1C:FB:84:7A:F9:D6:36:10:6F:42:82:55:4B:43:20:66:D1:64:CF:D0:76
a=ice-ufrag:2ad4320669f8831c24d2b0446ff212a4
a=ice-pwd:188bb68d2c0238fa3b95d9d65a5fa572
a=rtpmap:100 VP8/90000
a=sendrecv
a=rtcp-mux
a=ssrc:1087539294 cname:fd16395b-8bbb-493a-81e0-b2fde51244e0
a=msid:6b4a1e02-3eca-46dd-9fde-19117a147cbb 17003197-f770-465b-b1fb-9e71a8d790be
a=rtcp-fb:* transport-cc
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=rtcp-fb:* nack
a=extmap:1 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=mid:video-1

<--- Received SIP response (389 bytes) from WS:122.175.175.96:37529 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 10.0.11.55:8088;rport=8088;branch=z9hG4bKPj1173b431-c1ca-49d5-93fc-6fb09d95e266;alias
From: <sip:webrtc_client02@ip-10-0-11-55>;tag=4304a09f-efb4-4089-98bb-d270e4341fe7
To: <sips:webrtc_client03@122.175.175.96;rtcweb-breaker=no>
Call-ID: 1fdf6f6b-b778-4026-b7fc-b94f8f35e11f
CSeq: 2367 INVITE
Content-Length: 0


<--- Received SIP response (530 bytes) from WS:122.175.175.96:37529 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 10.0.11.55:8088;rport=8088;branch=z9hG4bKPj1173b431-c1ca-49d5-93fc-6fb09d95e266;alias
From: <sip:webrtc_client02@ip-10-0-11-55>;tag=4304a09f-efb4-4089-98bb-d270e4341fe7
To: <sips:webrtc_client03@122.175.175.96;rtcweb-breaker=no>;tag=MJ0pKgXMMF3I7XFex8lP
Contact: <sips:webrtc_client03@df7jal23ls0d.invalid;transport=wss>
Call-ID: 1fdf6f6b-b778-4026-b7fc-b94f8f35e11f
CSeq: 2367 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE


    -- PJSIP/webrtc_client03-000000c2 is ringing
    -- PJSIP/webrtc_client03-000000c2 is ringing
<--- Transmitting SIP response (572 bytes) to WS:122.175.175.96:43212 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=43212;received=122.175.175.96;branch=z9hG4bKEmIIJTrJUqxwP7QR4kQR0Mq0NUSf7TdF
Call-ID: 39997fb9-bff0-4fdc-dd5b-5a1fbfa1f0bd
From: <sip:webrtc_client02@sip.mysipdomain.com>;tag=O8nAOtKBaGyB9nDHVB9J
To: <sip:webrtc_client03@sip.mysipdomain.com>;tag=56c093a3-422d-42d2-94db-9bd5022afed4
CSeq: 17059 INVITE
Server: Asterisk PBX 16.4.1
Contact: <sips:10.0.11.55:8088;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Content-Length:  0


<--- Received SIP response (2141 bytes) from WS:122.175.175.96:37529 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 10.0.11.55:8088;rport=8088;branch=z9hG4bKPj1173b431-c1ca-49d5-93fc-6fb09d95e266;alias
From: <sip:webrtc_client02@ip-10-0-11-55>;tag=4304a09f-efb4-4089-98bb-d270e4341fe7
To: <sips:webrtc_client03@122.175.175.96;rtcweb-breaker=no>;tag=MJ0pKgXMMF3I7XFex8lP
Contact: <sips:webrtc_client03@df7jal23ls0d.invalid;transport=wss>
Call-ID: 1fdf6f6b-b778-4026-b7fc-b94f8f35e11f
CSeq: 2367 INVITE
Content-Type: application/sdp
Content-Length: 1582
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE

v=0
o=- 6427411359488548000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio-0 video-1
a=msid-semantic: WMS Ms541yEMrnbtZ2cY5Z6F3IvKuwBEwiD5ZI3A
m=audio 46898 UDP/TLS/RTP/SAVPF 0 101
c=IN IP4 192.168.1.2
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2437072876 1 udp 2122260223 192.168.1.2 46898 typ host generation 0 network-id 1 network-cost 10
a=candidate:3753982748 1 tcp 1518280447 192.168.1.2 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:fhNu
a=ice-pwd:6CfsQspSm8xGg2+MrgemUgQB
a=ice-options:trickle
a=fingerprint:sha-256 E5:87:96:28:CA:E8:5F:B1:13:BC:36:00:47:09:B4:DD:BD:99:68:E7:F0:D2:0D:52:FD:46:AB:6D:8B:45:92:7A
a=setup:active
a=mid:audio-0
a=sendrecv
a=msid:Ms541yEMrnbtZ2cY5Z6F3IvKuwBEwiD5ZI3A dcb81a41-8431-4359-8ccc-150476dae7d9
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ssrc:742794974 cname:qIZdE8ai0AMOHLO7
m=video 9 UDP/TLS/RTP/SAVPF 100
c=IN IP4 127.0.0.1
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:fhNu
a=ice-pwd:6CfsQspSm8xGg2+MrgemUgQB
a=ice-options:trickle
a=fingerprint:sha-256 E5:87:96:28:CA:E8:5F:B1:13:BC:36:00:47:09:B4:DD:BD:99:68:E7:F0:D2:0D:52:FD:46:AB:6D:8B:45:92:7A
a=setup:active
a=mid:video-1
a=extmap:1 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=sendrecv
a=msid:Ms541yEMrnbtZ2cY5Z6F3IvKuwBEwiD5ZI3A a2089d4d-b213-4eb7-92c1-2cc120c12ce7
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 goog-remb
a=rtcp-fb:100 transport-cc
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=ssrc:907548848 cname:qIZdE8ai0AMOHLO7

       > 0x7fcacc079580 -- Strict RTP learning after remote address set to: 192.168.1.2:46898
       > 0x7fcacc1b7100 -- Strict RTP learning after remote address set to: 192.168.1.2:46898
<--- Transmitting SIP request (472 bytes) to WS:122.175.175.96:37529 --->
ACK sips:webrtc_client03@122.175.175.96:37529;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.0.11.55:8088;rport;branch=z9hG4bKPj475d2345-b5f0-4967-9412-cfced4b5e28d;alias
From: <sip:webrtc_client02@ip-10-0-11-55>;tag=4304a09f-efb4-4089-98bb-d270e4341fe7
To: <sips:webrtc_client03@122.175.175.96;rtcweb-breaker=no>;tag=MJ0pKgXMMF3I7XFex8lP
Call-ID: 1fdf6f6b-b778-4026-b7fc-b94f8f35e11f
CSeq: 2367 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 16.4.1
Content-Length:  0


    -- PJSIP/webrtc_client03-000000c2 answered PJSIP/webrtc_client02-000000c1
       > 0x7fcad00d5930 -- Strict RTP learning after remote address set to: 192.168.1.4:44100
       > 0x7fcad00f5890 -- Strict RTP learning after remote address set to: 192.168.1.4:44100
<--- Transmitting SIP response (2327 bytes) to WS:122.175.175.96:43212 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=43212;received=122.175.175.96;branch=z9hG4bKEmIIJTrJUqxwP7QR4kQR0Mq0NUSf7TdF
Call-ID: 39997fb9-bff0-4fdc-dd5b-5a1fbfa1f0bd
From: <sip:webrtc_client02@sip.mysipdomain.com>;tag=O8nAOtKBaGyB9nDHVB9J
To: <sip:webrtc_client03@sip.mysipdomain.com>;tag=56c093a3-422d-42d2-94db-9bd5022afed4
CSeq: 17059 INVITE
Server: Asterisk PBX 16.4.1
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: <sips:10.0.11.55:8088;transport=ws>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:  1678

v=0
o=- 3097176616 4 IN IP4 10.0.11.55
s=Asterisk
c=IN IP4 10.0.11.55
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0 1
m=audio 13560 UDP/TLS/RTP/SAVPF 0 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 10:06:A5:A0:CA:52:B9:B6:D8:A7:56:A5:4E:9A:2B:45:15:53:A4:88:CE:C1:D6:4C:B1:25:D1:C7:30:57:CB:46
a=ice-ufrag:320b357c187486bf089c27930db85373
a=ice-pwd:605620c675c87d876c1849433dc18f93
a=candidate:H821e2c3e 1 UDP 2130706431 fe80::8f3:f5ff:fee0:bd36 13560 typ host
a=candidate:Ha000b37 1 UDP 2130706431 10.0.11.55 13560 typ host
a=candidate:S22f4393b 1 UDP 1694498815 34.244.57.59 13560 typ srflx raddr 10.0.11.55 rport 13560
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:1793814859 cname:85537ef4-4e80-49da-8a57-90b034c6abdf
a=msid:6d21591f-0d67-4b63-bfc0-d87f0e635716 1a2c386b-56f7-4bed-947d-230aea64ac4b
a=rtcp-fb:* transport-cc
a=mid:0
m=video 13560 UDP/TLS/RTP/SAVPF 96
a=connection:new
a=setup:active
a=fingerprint:SHA-256 10:06:A5:A0:CA:52:B9:B6:D8:A7:56:A5:4E:9A:2B:45:15:53:A4:88:CE:C1:D6:4C:B1:25:D1:C7:30:57:CB:46
a=ice-ufrag:320b357c187486bf089c27930db85373
a=ice-pwd:605620c675c87d876c1849433dc18f93
a=rtpmap:96 VP8/90000
a=sendrecv
a=rtcp-mux
a=ssrc:2096555701 cname:09fdeaa9-1913-4454-9d8c-92f356d51550
a=msid:6d21591f-0d67-4b63-bfc0-d87f0e635716 e27af025-86a0-49c4-bd75-fe59c8e3d642
a=rtcp-fb:* transport-cc
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=rtcp-fb:* nack
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=mid:1

    -- Channel PJSIP/webrtc_client03-000000c2 joined 'simple_bridge' basic-bridge <32279bbb-0bbf-4fa7-aebd-ab405033fc7e>
    -- Channel PJSIP/webrtc_client02-000000c1 joined 'simple_bridge' basic-bridge <32279bbb-0bbf-4fa7-aebd-ab405033fc7e>
[2020-04-23 21:15:17] ERROR[16340]: pjproject: <?>: 	   icess0x7fcad0162b88 ..Error sending STUN request: Network is unreachable
       > 0x7fcacc1b7100 -- Strict RTP learning after remote address set to: 122.175.175.96:46898
       > 0x7fcacc079580 -- Strict RTP learning after ICE completion
<--- Received SIP request (904 bytes) from WS:122.175.175.96:43212 --->
ACK sips:10.0.11.55:8088;transport=ws SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKZxuAXf7MmTM8qBVQwgyF;rport
From: <sip:webrtc_client02@sip.mysipdomain.com>;tag=O8nAOtKBaGyB9nDHVB9J
To: <sip:webrtc_client03@sip.mysipdomain.com>;tag=56c093a3-422d-42d2-94db-9bd5022afed4
Contact: <sips:webrtc_client02@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 39997fb9-bff0-4fdc-dd5b-5a1fbfa1f0bd
CSeq: 17059 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="webrtc_client02",realm="asterisk",nonce="1587676512/937cab544e0ef68c9279cdf90d7a9913",uri="sips:10.0.11.55:8088;transport=ws",response="2df5eca684b6d2c27ff78cf2976ddf5c",algorithm=md5,cnonce="da32e7ef04ef4a36eebeb333f35deece",opaque="51aac44b1493c5eb",qop=auth,nc=00000002
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom


       > 0x7fcad00d5930 -- Strict RTP learning after remote address set to: 122.175.175.96:44100
       > 0x7fcad00f5890 -- Strict RTP learning after ICE completion
       > 0x7fcad00f5890 -- Strict RTP switching to RTP target address 122.175.175.96:44100 as source
       > 0x7fcad00d5930 -- Strict RTP switching to RTP target address 122.175.175.96:44100 as source
[2020-04-23 21:15:18] NOTICE[13379][C-00000067]: res_rtp_asterisk.c:6663 rtp_instance_parse_transport_wide_cc: Starting feedback
       > 0x7fcacc079580 -- Strict RTP switching to RTP target address 122.175.175.96:46898 as source
       > 0x7fcacc1b7100 -- Strict RTP switching to RTP target address 122.175.175.96:46898 as source
[2020-04-23 21:15:19] NOTICE[13385][C-00000067]: res_rtp_asterisk.c:6663 rtp_instance_parse_transport_wide_cc: Starting feedback
       > 0x7fcacc079580 -- Strict RTP learning complete - Locking on source address 122.175.175.96:46898
       > 0x7fcacc1b7100 -- Strict RTP learning complete - Locking on source address 122.175.175.96:46898
       > 0x7fcad00d5930 -- Strict RTP learning complete - Locking on source address 122.175.175.96:44100
       > 0x7fcad00f5890 -- Strict RTP learning complete - Locking on source address 122.175.175.96:44100
<--- Received SIP request (854 bytes) from WS:122.175.175.96:43212 --->
BYE sips:10.0.11.55:8088;transport=ws SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKCGAfXBu1s2hfprks05N4GQexVimfkwSC;rport
From: <sip:webrtc_client02@sip.mysipdomain.com>;tag=O8nAOtKBaGyB9nDHVB9J
To: <sip:webrtc_client03@sip.mysipdomain.com>;tag=56c093a3-422d-42d2-94db-9bd5022afed4
Call-ID: 39997fb9-bff0-4fdc-dd5b-5a1fbfa1f0bd
CSeq: 17060 BYE
Content-Length: 0
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;language="en,fr"
Authorization: Digest username="webrtc_client02",realm="asterisk",nonce="1587676512/937cab544e0ef68c9279cdf90d7a9913",uri="sips:10.0.11.55:8088;transport=ws",response="804e9d2b95719674f192769017f2c9f8",algorithm=md5,cnonce="da32e7ef04ef4a36eebeb333f35deece",opaque="51aac44b1493c5eb",qop=auth,nc=00000003
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom


<--- Transmitting SIP response (407 bytes) to WS:122.175.175.96:43212 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=43212;received=122.175.175.96;branch=z9hG4bKCGAfXBu1s2hfprks05N4GQexVimfkwSC
Call-ID: 39997fb9-bff0-4fdc-dd5b-5a1fbfa1f0bd
From: <sip:webrtc_client02@sip.mysipdomain.com>;tag=O8nAOtKBaGyB9nDHVB9J
To: <sip:webrtc_client03@sip.mysipdomain.com>;tag=56c093a3-422d-42d2-94db-9bd5022afed4
CSeq: 17060 BYE
Server: Asterisk PBX 16.4.1
Content-Length:  0


    -- Channel PJSIP/webrtc_client02-000000c1 left 'simple_bridge' basic-bridge <32279bbb-0bbf-4fa7-aebd-ab405033fc7e>
    -- Channel PJSIP/webrtc_client03-000000c2 left 'simple_bridge' basic-bridge <32279bbb-0bbf-4fa7-aebd-ab405033fc7e>
  == Spawn extension (internal, webrtc_client03, 2) exited non-zero on 'PJSIP/webrtc_client02-000000c1'
<--- Transmitting SIP request (496 bytes) to WS:122.175.175.96:37529 --->
BYE sips:webrtc_client03@122.175.175.96:37529;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.0.11.55:8088;rport;branch=z9hG4bKPj3267c7c9-0b18-48a2-80e2-261fc100ab3d;alias
From: <sip:webrtc_client02@ip-10-0-11-55>;tag=4304a09f-efb4-4089-98bb-d270e4341fe7
To: <sips:webrtc_client03@122.175.175.96;rtcweb-breaker=no>;tag=MJ0pKgXMMF3I7XFex8lP
Call-ID: 1fdf6f6b-b778-4026-b7fc-b94f8f35e11f
CSeq: 2368 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 16.4.1
Content-Length:  0


<--- Received SIP response (441 bytes) from WS:122.175.175.96:37529 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 10.0.11.55:8088;rport=8088;branch=z9hG4bKPj3267c7c9-0b18-48a2-80e2-261fc100ab3d;alias
From: <sip:webrtc_client02@ip-10-0-11-55>;tag=4304a09f-efb4-4089-98bb-d270e4341fe7
To: <sips:webrtc_client03@122.175.175.96;rtcweb-breaker=no>;tag=MJ0pKgXMMF3I7XFex8lP
Contact: <sips:webrtc_client03@df7jal23ls0d.invalid;transport=wss>
Call-ID: 1fdf6f6b-b778-4026-b7fc-b94f8f35e11f
CSeq: 2368 BYE
Content-Length: 0
4

I didn’t see any private ips in rtp debug, the public ip with kust the two ports the remote address is being set to in the above logs. Couldn’t see anything specific to audio or video streams. If I’m correct in thinking that type 00 refers to ulaw then it seems to be sent to both locations.
The following is a snippet from another call I tried.

Sent RTP packet to      122.175.175.96:46554 (via ICE) (type 96, seq 017498, ts 398211894, len 000392)
Sent RTP packet to      122.175.175.96:46554 (via ICE) (type 96, seq 017499, ts 398215224, len 000436)
Sent RTP packet to      122.175.175.96:46554 (via ICE) (type 96, seq 017500, ts 398218554, len 000334)
Got  RTP packet from    122.175.175.96:48692 (type 00, seq 016848, ts 2480405775, len 000170)
Sent RTP packet to      122.175.175.96:46554 (via ICE) (type 00, seq 012425, ts 2480405776, len 000170)
       > 0x7fcac42ddf20 -- Strict RTP learning complete - Locking on source address 122.175.175.96:46554
Got  RTP packet from    122.175.175.96:46554 (type 96, seq 010313, ts 3293665773, len 000566)
Got  RTP packet from    122.175.175.96:48692 (type 00, seq 016849, ts 2480405935, len 000170)
Sent RTP packet to      122.175.175.96:46554 (via ICE) (type 00, seq 012426, ts 2480405936, len 000170)
Got  RTP packet from    122.175.175.96:46554 (type 00, seq 006743, ts 712609440, len 000170)
Sent RTP packet to      122.175.175.96:48692 (via ICE) (type 00, seq 023592, ts 712609440, len 000170)
Got  RTP packet from    122.175.175.96:48692 (type 100, seq 013357, ts 398221794, len 000355)
Sent RTP packet to      122.175.175.96:46554 (via ICE) (type 96, seq 017501, ts 398221794, len 000347)
Got  RTP packet from    122.175.175.96:46554 (type 00, seq 006744, ts 712609600, len 000170)
Sent RTP packet to      122.175.175.96:48692 (via ICE) (type 00, seq 023593, ts 712609600, len 000170)
Got  RTP packet from    122.175.175.96:48692 (type 00, seq 016850, ts 2480406095, len 000170)
Sent RTP packet to      122.175.175.96:46554 (via ICE) (type 00, seq 012427, ts 2480406096, len 000170)
Got  RTP packet from    122.175.175.96:48692 (type 00, seq 016851, ts 2480406255, len 000170)
Sent RTP packet to      122.175.175.96:46554 (via ICE) (type 00, seq 012428, ts 2480406256, len 000170)
Got  RTP packet from    122.175.175.96:46554 (type 00, seq 006746, ts 712609920, len 000170)
Sent RTP packet to      122.175.175.96:48692 (via ICE) (type 00, seq 023595, ts 712609920, len 000170)
Got  RTP packet from    122.175.175.96:48692 (type 100, seq 013358, ts 398225124, len 000383)
Sent RTP packet to      122.175.175.96:46554 (via ICE) (type 96, seq 017502, ts 398225124, len 000375)
Got  RTP packet from    122.175.175.96:48692 (type 00, seq 016852, ts 2480406415, len 000170)
Sent RTP packet to      122.175.175.96:46554 (via ICE) (type 00, seq 012429, ts 2480406416, len 000170)
Got  RTP packet from    122.175.175.96:46554 (type 00, seq 006748, ts 712610240, len 000170)
Sent RTP packet to      122.175.175.96:48692 (via ICE) (type 00, seq 023597, ts 712610240, len 000170)
Got  RTP packet from    122.175.175.96:48692 (type 00, seq 016853, ts 2480406575, len 000170)
Sent RTP packet to      122.175.175.96:46554 (via ICE) (type 00, seq 012430, ts 2480406576, len 000170)
Got  RTP packet from    122.175.175.96:46554 (type 00, seq 006750, ts 712610560, len 000170)
Sent RTP packet to      122.175.175.96:48692 (via ICE) (type 00, seq 023599, ts 712610560, len 000170)
5

Is the SIP trace, and rtp debug from the same call? The SIP trace shows ICE negotiating private addresses, and different ports than where audio is coming from/going to in the rtp debug.

Also according to the rtp debug audio is coming from and going to the same IP address, but different ports. Unless everything is hosted on the same machine, which based on your description it doesn’t sound that way, then that would be a problem.

And yes. 00 is ulaw.

6

Thanks for responding. I had my doubts about that private ip in ICE negotiation, but I found it pretty odd that video was working fine in that case. I’ve tried many things but no progress so far.

yes the rtp debug was from a different call…i took a separate trace so as to not mess the logs up.
As I mentioned with the sip trace, the two client devices are on the same network, connected to asterisk server(hosted on AWS) using sipml5. Following is both sip and rtp traces in one. I added google’s stun server this time in sipml5. Results are still the same.
Audio only calls seem to work, albeit with poor quality and sensitive loopback issues

https://pastebin.com/qR6nsCvW

Please take a look at it.

7

From what I can tell it looks like stuff negotiated okay. It also appears that Asterisk is both sending and receiving audio, and video to the negotiated addresses/ports. Make sure no ports are being blocked on the browser side.

There is packet loss, but it’s hard to tell how severe as it appears some of the logging has been removed.

Since it looks like Asterisk is sending/receiving the audio you might next check the browser logs and ensure it’s receiving the expected audio. You could also get a pcap. The packets will be encrypted but you should be able to at least tell where they are coming/going, and if you label them as RTP (say in wireshark) it’ll even give you a little more information.

I did also note a couple things in the log :

[2020-04-24 22:00:52] ERROR[4601]: pjproject: <?>: icess0x7f4500028918 …Error sending STUN request: Network is unreachable

And

[2020-04-24 22:01:00] WARNING[6162][C-00000006]: res_srtp.c:493 ast_srtp_protect: SRTP protect: replay check failed (index too old)

I’m not sure if those would be related, but it could be indicative of some network and/or srtp problems. Ensure you’re using a supported version of libsrtp for instance.

1 Like
8

Hi @kharwell, I’m looking into the client side traces.

Yes, I removed the RTP logging for readability.

Both libsrtp 1.4.5 and 2.1.0 are installed on our Ubuntu server, and they both are not exactly recommended it seems. Since pjsip is also bundled with a version, is there any way to make sure which version is being used by asterisk? I looked and couldn’t find anything on this

9

If you didn’t specify anything for the “configure” option then Asterisk will use what’s installed in the default location. For a different location or custom install you can tell Asterisk where to find it when configuring. See

./configure --help

For more information. For instance:

–with-srtp=PATH use Secure RTP files in PATH

10

Another thing to try is if you can setup Asterisk locally, and on the same network as the endpoints. If you can get everything working locally first then it’d point to something network related.

11

Should be network configuration related I think, 'cause it works when we don’t enable webrtc in config and use udp transport directly using linphone. In case of webrtc we are terminating tls on the load balancer itself. This is the updated config. Had to disable dtls related settings as there were some errors, I think because it was trying to decrypt already decrypted traffic.

[webrtc_client02]
type=endpoint
aors=webrtc_client02
auth=webrtc_client02
webrtc=yes
context=internal
disallow=all
allow=ulaw,vp9
force_rport=yes
direct_media=no
media_encryption=no
dtls_verify=No
dtls_setup=active
rtp_symmetric=yes
12

Setting “webrtc=yes” is a shortcut for setting required defaults for a few of those settings. For example “media_encryption=dtls”. I’m not sure what happens if you set both like you have.

You can find more information here:

https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients

13

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.

15

Try this: https://www.innovateasterisk.com/phone/

If it works for you, there is more info here:

3 Likes
16

As @kharwell suggested, it was indeed an issue with the client library. Demos for both JSSip and SIPML5 are probably outdated. Debugging using chrome://webrtc-internals we found out that an audio stream was being received, but not being played. Messing around with our implementation based on jssip worked :). We also faced issues with calls from webrtc client to android sip clients, zoiper(call dropped as soon as it arrived) as well as linphone(wasn’t negotiating ICE). Android to webrtc client was working already. This too turned out to be client issue. Using Antisip on android worked.
Thanks a lot for your support.

@InnovateAsterisk looks good, will try it out :slight_smile:

17

Stream received - but not played since it could have different source IP ( not same as in SDP)

Try to install ICE ( STUN+TURN) server on your asterisk server.
and specify javascript library you are using to use ice servers, smth line this (sipjs ) :slight_smile:

I have this config (sipjs + asterisk over WEBrtc (wss with valid ssl cert! ) ) working almost in 99% if cases

Please note - by specifig user/pass for ICE server, you automatically enable TURN function, which proxy RTP )

 userAgent = new SIP.UA({
                          uri: wuser +'@'+serverip,
                          authorizationUser: wuser,
                          password: wpass ,
                          transportOptions: {
                              wsServers: [ 'wss://' + serverip + ':8443/ws' ],
                              maxReconnectionAttempts: 999,
                              reconnectionTimeout: 3,
                              keepAliveInterval: 2,
                              traceSip: true
                          },
                          userAgentString: 'Cloud CRM ',
                          register: true,
                          expires: 15,
                          rtcpMuxPolicy: 'yes',
                          hackIpInContact: true,
                          hackWssInTransport: true,
                          stunServers: ["stun:serverip:19302"],

                          turnServers: [{
                                          urls: "turn:serverip:19302?transport=udp",
                                          username: "turn",
                                          password: "secured",
                                          credential: "secured"
                          }],

                          // Acoording to :  https://groups.google.com/forum/#!topic/sip_js/VYMi4UMbxvM
                          sessionDescriptionHandlerFactoryOptions: {
                            peerConnectionOptions: {
                              iceCheckingTimeout:6000,
                                  rtcConfiguration:{
                                         iceTransportPolicy: "relay",
                                     iceServers:
                                                [
                                                  { urls:"stun:" + serverip + ":19302" },
                                                  {
                                                    urls: "turn:serverip:19302?transport=udp",
                                                        username: "turn",
                                                        credential: "secured"

                                                  }
                                                ]
                              }
                            }
                          }

                });
1 Like