[asterisk-users] PJSIP - Video Support for WebRTC

Wed Jul 27 07:35:41 CDT 2016

On 07/26/2016 03:15 PM, Olivier wrote:
> Matthew Jordan <mjordan <at> digium.com> writes:
>
>>
>> On Mon, Mar 23, 2015 at 8:55 AM, Gosmac <goseeped <at> gmail.com>
> wrote:
>>> Hey i have an interesting topic to discuss here.
>>>
>>> The main goal here is to be able to make a video call between two
> WebRTC endpoints registered on asterisk 13
>> it is a feature that definitely asterisk 13 should support .
>>>
>>> the problems that i faced with this is the following and i hope i
> could get an advise here.
>>>
>>> asterisk 13 vanilla version has some issues marking the video
> packets this complain web browser
>> specially VP8 codecs so a friend of mine help me to patch
> res_rtp_asterisk and now asterisk is marking
>> video streams :) it just mark video packets not touch anything else
> and web browser show video on web page
>> now I’m using online demo http://tryit.jssip.net/ is stable and get
> more updates than sipml5. so i try
>> echo() dialplan test and everything work perfect on echo test :).
>>>
>>> i have two questions and i hope you could give me some advise.
>>>
>>> 1) after marking video packet I’m able to make Dial() between two
> webrtc peers but i get one way audio and
>> video on callee party, “after 3 minutes on call” i get two way audio
> and video on all parties seems to be
>> not just a problem on a missing keyframe.
>>>
>>>  1.1) the 3 minutes delay only happen using chrome stable , could be
> a dtls problem when asterisk make an
>> offer to other endpoint?
>>>  1.2) when i use chrome-dev and i disable dlts encryption everything
> work perfect on video call.
>>>
>>> 2) after marking video packets i realize that when you make a call
> with video and you involve on dialplan an
>> application like playback or music on hold any application that
> played audio files (audio and video never work).
>>>
>>> 2.1) asterisk is muggling the audio and video streams ?
>>>
>>> This is good information for all guys out there that wants to
> support video on webrtc in asterisk 13
>>>
>>
>> Please stop spamming the list with this e-mail. Resending it multiple
>> times is clearly not yielding the results you'd like.
>>
>
> Hi Matthew,
> I'm testing WebRTC (JSSIP) with Asterisk 12.8 after following the
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support link.
> Using Firefox, I can connect both JSSIP Clients to asterisk. When I Call
> one Client, the Client just Ring One Time and after pick up a receive
> WebRTC error on the Firefox browser.
> Here is my asterisk sip debug:
>
> <--- SIP read from WS:192.168.2.103:49851 ---> INVITE
> sip:6000 at 192.168.2.106 SIP/2.0
> Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK8394689
> Max-Forwards: 69
> To: <sip:6000 at 192.168.2.106>
> From: "6001" <sip:6001 at 192.168.2.106>;tag=m6bqn333dr
> Call-ID: 1ansppdrpdulbtr3j5ub
> CSeq: 6407 INVITE
> X-Can-Renegotiate: false
> Contact: <sip:5krseuop at 0iemcrsq9tm0.invalid;transport=ws;ob>
> Content-Type: application/sdp
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
> Supported: ice,replaces,outbound
> User-Agent: JsSIP 2.0.2
> Content-Length: 3158
>
> v=0
> o=mozilla...THIS_IS_SDPARTA-47.0.1 5760840281459352758 0 IN IP4 0.0.0.0
> s=-
> t=0 0
> a=sendrecv
> a=fingerprint:sha-256
> E8:B7:2A:C2:DF:8B:AA:74:E6:D6:93:1C:68:88:81:39:82:C2:31:45:3D:8C:23:DF:
> C1:23:72:03:F6:61:CC:F6
> a=group:BUNDLE sdparta_0 sdparta_1
> a=ice-options:trickle
> a=msid-semantic:WMS *
> m=audio 56808 UDP/TLS/RTP/SAVPF 109 9 0 8 c=IN IP4 87.169.189.102
> a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
> 56806 typ host
> a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
> 56807 typ host
> a=candidate:4 1 UDP 2122055935 192.168.2.103 56808 typ host
> a=candidate:6 1 UDP 2122252543 192.168.56.1 56809 typ host a=candidate:0
> 2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a
> 56810 typ host
> a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc:43fa
> 56811 typ host
> a=candidate:4 2 UDP 2122055934 192.168.2.103 56812 typ host
> a=candidate:6 2 UDP 2122252542 192.168.56.1 56813 typ host
> a=candidate:5 1 UDP 1685856255 87.169.189.102 56808 typ srflx raddr
> 192.168.2.103 rport 56808
> a=candidate:5 2 UDP 1685856254 87.169.189.102 56812 typ srflx raddr
> 192.168.2.103 rport 56812
> a=sendrecv
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=fmtp:109 maxplaybackrate=48000;stereo=1 a=ice-
> pwd:138f583004cb3079134e8e8f20dac36f
> a=ice-ufrag:0941ac54
> a=mid:sdparta_0
> a=msid:{fb724d76-44fe-4e7d-a8d8-e4c00b4b57fe}
> {bba6da45-42c8-4529-8f4b-046cffcdc40d}
> a=rtcp:56812 IN IP4 87.169.189.102
> a=rtcp-mux
> a=rtpmap:109 opus/48000/2
> a=rtpmap:9 G722/8000/1
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=setup:actpass
> a=ssrc:540714091 cname:{ecde75c0-993f-44af-b136-8944915fe31c}
> m=video 56816 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 87.169.189.102
> a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
> 56814 typ host
> a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
> 56815 typ host
> a=candidate:4 1 UDP 2122055935 192.168.2.103 56816 typ host
> a=candidate:6 1 UDP 2122252543 192.168.56.1 60290 typ host a=candidate:0
> 2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a
> 60291 typ host
> a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc:43fa
> 60292 typ host
> a=candidate:4 2 UDP 2122055934 192.168.2.103 64504 typ host
> a=candidate:6 2 UDP 2122252542 192.168.56.1 64505 typ host
> a=candidate:5 1 UDP 1685856255 87.169.189.102 56816 typ srflx raddr
> 192.168.2.103 rport 56816
> a=candidate:5 2 UDP 1685856254 87.169.189.102 64504 typ srflx raddr
> 192.168.2.103 rport 64504
> a=recvonly
> a=fmtp:120 max-fs=12288;max-fr=60
> a=fmtp:126
> profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1
> a=fmtp:97 profile-level-id=42e01f;level-asymmetry-allowed=1
> a=ice-pwd:138f583004cb3079134e8e8f20dac36f
> a=ice-ufrag:0941ac54
> a=mid:sdparta_1
> a=rtcp:64504 IN IP4 87.169.189.102
> a=rtcp-fb:120 nack
> a=rtcp-fb:120 nack pli
> a=rtcp-fb:120 ccm fir
> a=rtcp-fb:126 nack
> a=rtcp-fb:126 nack pli
> a=rtcp-fb:126 ccm fir
> a=rtcp-fb:97 nack
> a=rtcp-fb:97 nack pli
> a=rtcp-fb:97 ccm fir
> a=rtcp-mux
> a=rtpmap:120 VP8/90000
> a=rtpmap:126 H264/90000
> a=rtpmap:97 H264/90000
> a=setup:actpass
> a=ssrc:3124982 cname:{ecde75c0-993f-44af-b136-8944915fe31c}
> <------------->
> --- (14 headers 71 lines) ---
> Using INVITE request as basis request - 1ansppdrpdulbtr3j5ub Found peer
> '6001' for '6001' from 192.168.2.103:49851
>
> <--- Reliably Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 401
> Unauthorized
> Via: SIP/2.0/WS
> 0iemcrsq9tm0.invalid;branch=z9hG4bK8394689;received=192.168.2.103;rport=
> 49851
> From: "6001" <sip:6001 at 192.168.2.106>;tag=m6bqn333dr
> To: <sip:6000 at 192.168.2.106>;tag=as28bc71f3
> Call-ID: 1ansppdrpdulbtr3j5ub
> CSeq: 6407 INVITE
> Server: Asterisk PBX 12.8.2
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="192.168.2.106",
> nonce="69dcc467"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog '1ansppdrpdulbtr3j5ub' in 32000 ms
> (Method: INVITE)
>
> <--- SIP read from WS:192.168.2.103:49851 ---> ACK
> sip:6000 at 192.168.2.106 SIP/2.0
> Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK8394689
> To: <sip:6000 at 192.168.2.106>;tag=as28bc71f3
> From: "6001" <sip:6001 at 192.168.2.106>;tag=m6bqn333dr
> Call-ID: 1ansppdrpdulbtr3j5ub
> CSeq: 6407 ACK
> Content-Length: 0
>
> <------------->
> --- (7 headers 0 lines) ---
>
> <--- SIP read from WS:192.168.2.103:49851 ---> INVITE
> sip:6000 at 192.168.2.106 SIP/2.0
> Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK7826783
> Max-Forwards: 69
> To: <sip:6000 at 192.168.2.106>
> From: "6001" <sip:6001 at 192.168.2.106>;tag=m6bqn333dr
> Call-ID: 1ansppdrpdulbtr3j5ub
> CSeq: 6408 INVITE
> Authorization: Digest algorithm=MD5, username="6001",
> realm="192.168.2.106", nonce="69dcc467", uri="sip:6000 at 192.168.2.106",
> response="845417814e71ce56e93d846538ea31ae"
> X-Can-Renegotiate: false
> Contact: <sip:5krseuop at 0iemcrsq9tm0.invalid;transport=ws;ob>
> Content-Type: application/sdp
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
> Supported: ice,replaces,outbound
> User-Agent: JsSIP 2.0.2
> Content-Length: 3158
>
> v=0
> o=mozilla...THIS_IS_SDPARTA-47.0.1 5760840281459352758 0 IN IP4 0.0.0.0
> s=-
> t=0 0
> a=sendrecv
> a=fingerprint:sha-256
> E8:B7:2A:C2:DF:8B:AA:74:E6:D6:93:1C:68:88:81:39:82:C2:31:45:3D:8C:23:DF:
> C1:23:72:03:F6:61:CC:F6
> a=group:BUNDLE sdparta_0 sdparta_1
> a=ice-options:trickle
> a=msid-semantic:WMS *
> m=audio 56808 UDP/TLS/RTP/SAVPF 109 9 0 8 c=IN IP4 87.169.189.102
> a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
> 56806 typ host
> a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
> 56807 typ host
> a=candidate:4 1 UDP 2122055935 192.168.2.103 56808 typ host
> a=candidate:6 1 UDP 2122252543 192.168.56.1 56809 typ host a=candidate:0
> 2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a
> 56810 typ host
> a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc:43fa
> 56811 typ host
> a=candidate:4 2 UDP 2122055934 192.168.2.103 56812 typ host
> a=candidate:6 2 UDP 2122252542 192.168.56.1 56813 typ host
> a=candidate:5 1 UDP 1685856255 87.169.189.102 56808 typ srflx raddr
> 192.168.2.103 rport 56808
> a=candidate:5 2 UDP 1685856254 87.169.189.102 56812 typ srflx raddr
> 192.168.2.103 rport 56812
> a=sendrecv
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=fmtp:109 maxplaybackrate=48000;stereo=1 a=ice-
> pwd:138f583004cb3079134e8e8f20dac36f
> a=ice-ufrag:0941ac54
> a=mid:sdparta_0
> a=msid:{fb724d76-44fe-4e7d-a8d8-e4c00b4b57fe}
> {bba6da45-42c8-4529-8f4b-046cffcdc40d}
> a=rtcp:56812 IN IP4 87.169.189.102
> a=rtcp-mux
> a=rtpmap:109 opus/48000/2
> a=rtpmap:9 G722/8000/1
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=setup:actpass
> a=ssrc:540714091 cname:{ecde75c0-993f-44af-b136-8944915fe31c}
> m=video 56816 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 87.169.189.102
> a=candidate:0 1 UDP 2122187007 2003:88:6908:7659:3dbc:5101:33ff:d07a
> 56814 typ host
> a=candidate:2 1 UDP 2122121471 2003:88:6908:7659:6113:7316:1ebc:43fa
> 56815 typ host
> a=candidate:4 1 UDP 2122055935 192.168.2.103 56816 typ host
> a=candidate:6 1 UDP 2122252543 192.168.56.1 60290 typ host a=candidate:0
> 2 UDP 2122187006 2003:88:6908:7659:3dbc:5101:33ff:d07a
> 60291 typ host
> a=candidate:2 2 UDP 2122121470 2003:88:6908:7659:6113:7316:1ebc:43fa
> 60292 typ host
> a=candidate:4 2 UDP 2122055934 192.168.2.103 64504 typ host
> a=candidate:6 2 UDP 2122252542 192.168.56.1 64505 typ host
> a=candidate:5 1 UDP 1685856255 87.169.189.102 56816 typ srflx raddr
> 192.168.2.103 rport 56816
> a=candidate:5 2 UDP 1685856254 87.169.189.102 64504 typ srflx raddr
> 192.168.2.103 rport 64504
> a=recvonly
> a=fmtp:120 max-fs=12288;max-fr=60
> a=fmtp:126
> profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1
> a=fmtp:97 profile-level-id=42e01f;level-asymmetry-allowed=1
> a=ice-pwd:138f583004cb3079134e8e8f20dac36f
> a=ice-ufrag:0941ac54
> a=mid:sdparta_1
> a=rtcp:64504 IN IP4 87.169.189.102
> a=rtcp-fb:120 nack
> a=rtcp-fb:120 nack pli
> a=rtcp-fb:120 ccm fir
> a=rtcp-fb:126 nack
> a=rtcp-fb:126 nack pli
> a=rtcp-fb:126 ccm fir
> a=rtcp-fb:97 nack
> a=rtcp-fb:97 nack pli
> a=rtcp-fb:97 ccm fir
> a=rtcp-mux
> a=rtpmap:120 VP8/90000
> a=rtpmap:126 H264/90000
> a=rtpmap:97 H264/90000
> a=setup:actpass
> a=ssrc:3124982 cname:{ecde75c0-993f-44af-b136-8944915fe31c}
> <------------->
> --- (15 headers 71 lines) ---
> Using INVITE request as basis request - 1ansppdrpdulbtr3j5ub Found peer
> '6001' for '6001' from 192.168.2.103:49851
>    == TLS/SSL ECDH initialized (secp256r1), faster PFS cipher-suites
> enabled
>    == TLS/SSL ECDH initialized (secp256r1), faster PFS cipher-suites
> enabled
>    == Using SIP RTP CoS mark 5
> Found RTP audio format 109
> Found RTP audio format 9
> Found RTP audio format 0
> Found RTP audio format 8
> Found audio description format opus for ID 109 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 RTP video format 120 Found
> RTP video format 126 Found RTP video format 97 Found video description
> format VP8 for ID 120 Found video description format H264 for ID 126
> Found video description format H264 for ID 97
> Capabilities: us -
> (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aa
> l2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren
> 14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin19
> 2|opus|vp8|silk8|silk12|silk16|silk24),
> peer - audio=(ulaw|alaw|g722|opus)/video=(h264|vp8)/text=(nothing),
> combined - (ulaw|alaw|g722|h264|opus|vp8) Non-codec capabilities (dtmf):
> us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0
> (nothing) Peer audio RTP is at port 87.169.189.102:56808 Peer doesn't
> provide T.140 Looking for 6000 in outgoing (domain 192.168.2.106)
> list_route: route/path hop:
> <sip:5krseuop at 0iemcrsq9tm0.invalid;transport=ws;ob>
>
> <--- Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 100 Trying
> Via: SIP/2.0/WS
> 0iemcrsq9tm0.invalid;branch=z9hG4bK7826783;received=192.168.2.103;rport=
> 49851
> From: "6001" <sip:6001 at 192.168.2.106>;tag=m6bqn333dr
> To: <sip:6000 at 192.168.2.106>
> Call-ID: 1ansppdrpdulbtr3j5ub
> CSeq: 6408 INVITE
> Server: Asterisk PBX 12.8.2
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:6000 at 192.168.2.106:5060;transport=WS>
> Content-Length: 0
>
>
> <------------>
>      -- Executing [6000 at outgoing:1] Dial("SIP/6001-00000000",
> "SIP/6000") in new stack
>    == TLS/SSL ECDH initialized (secp256r1), faster PFS cipher-suites
> enabled
>    == TLS/SSL ECDH initialized (secp256r1), faster PFS cipher-suites
> enabled
>    == Using SIP RTP CoS mark 5
> We think we can do text
> And we have a text rtp object
> Audio is at 17504
> Lets set up the text sdp
> Text is at 0.0.0.0:12836
> Adding codec 100003 (ulaw) to SDP
> Adding codec 100001 (g723) to SDP
> Adding codec 100002 (gsm) to SDP
> Adding codec 100004 (alaw) to SDP
> Adding codec 100011 (g726) to SDP
> Adding codec 100006 (adpcm) to SDP
> Adding codec 100007 (lpc10) to SDP
> Adding codec 100008 (g729) to SDP
> Adding codec 100009 (speex) to SDP
> Adding codec 100016 (speex16) to SDP
> Adding codec 100010 (ilbc) to SDP
> Adding codec 100005 (g726aal2) to SDP
> Adding codec 100012 (g722) to SDP
> Adding codec 100021 (slin16) to SDP
> Adding text codec 400001 (red) to SDP
> Adding text codec 400002 (t140) to SDP
> Adding codec 100013 (siren7) to SDP
> Adding codec 100014 (siren14) to SDP
> Adding codec 100017 (testlaw) to SDP
> Adding codec 100015 (g719) to SDP
> Adding codec 100028 (speex32) to SDP
> Adding codec 100020 (slin12) to SDP
> Adding codec 100022 (slin24) to SDP
> Adding codec 100023 (slin32) to SDP
> Adding codec 100024 (slin44) to SDP
> Adding codec 100025 (slin48) to SDP
> Adding codec 100026 (slin96) to SDP
> Adding codec 100027 (slin192) to SDP
> Adding codec 100030 (opus) to SDP
> Adding codec 100018 (silk8) to SDP
> Adding codec 100018 (silk12) to SDP
> Adding codec 100018 (silk16) to SDP
> Adding codec 100018 (silk24) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting
> (NAT) to 192.168.2.103:49848:
> INVITE sip:bbglnljp at 72rvpk435t95.invalid;transport=ws SIP/2.0
> Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
> Max-Forwards: 70
> From: "6001" <sip:6001 at 192.168.2.106>;tag=as0ba3cd59
> To: <sip:bbglnljp at 72rvpk435t95.invalid;transport=ws>
> Contact: <sip:6001 at 192.168.2.106:5060;transport=WS>
> Call-ID: 5681400a771147ed0c16fff2363c7e55 at 192.168.2.106:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 12.8.2
> Date: Sun, 24 Jul 2016 22:21:12 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 2778
>
> v=0
> o=root 1723742189 1723742189 IN IP4 192.168.2.106 s=Asterisk PBX 12.8.2
> c=IN IP4 192.168.2.106 t=0 0 m=audio 17504 RTP/SAVPF 0 4 3 8 111 5 7 18
> 110 117 97 112 9 118 102 115
> 116 119 107 96 108 109 113 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:4 G723/8000
> a=fmtp:4 annexa=no
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:5 DVI4/8000
> a=rtpmap:7 LPC/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:110 speex/8000
> a=rtpmap:117 speex/16000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:112 AAL2-G726-32/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:118 L16/16000
> a=rtpmap:102 G7221/16000
> a=fmtp:102 bitrate=32000
> a=rtpmap:115 G7221/32000
> a=fmtp:115 bitrate=48000
> a=rtpmap:116 G719/48000
> a=fmtp:116 bitrate=64000
> a=rtpmap:119 speex/32000
> a=rtpmap:107 opus/48000/2
> a=fmtp:107
> maxplaybackrate=48000;sprop-
> maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-
> stereo=0;cbr=0;useinbandfec=0;usedtx=0
> a=rtpmap:96 SILK/8000
> a=fmtp:96 maxaveragebitrate=10000
> a=fmtp:96 usedtx=0
> a=fmtp:96 useinbandfec=1
> a=rtpmap:108 SILK/12000
> a=fmtp:108 maxaveragebitrate=12000
> a=fmtp:108 usedtx=0
> a=fmtp:108 useinbandfec=1
> a=rtpmap:109 SILK/16000
> a=fmtp:109 maxaveragebitrate=20000
> a=fmtp:109 usedtx=0
> a=fmtp:109 useinbandfec=1
> a=rtpmap:113 SILK/24000
> a=fmtp:113 maxaveragebitrate=30000
> a=fmtp:113 usedtx=0
> a=fmtp:113 useinbandfec=1
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:20
> a=ice-ufrag:29c82286064fb054206fb5686f954dde
> a=ice-pwd:668a84152d94daf26a719ce95a2174fe
> a=candidate:Hc0a8026a 1 UDP 2130706431 192.168.2.106 17504 typ host
> a=candidate:S57a9bd66 1 UDP 1694498815 87.169.189.102 17504 typ srflx
> raddr 192.168.2.106 rport 17504 a=candidate:Hc0a8026a 2 UDP 2130706430
> 192.168.2.106 17505 typ host
> a=candidate:S57a9bd66 2 UDP 1694498814 87.169.189.102 17505 typ srflx
> raddr 192.168.2.106 rport 17505 a=connection:new a=setup:actpass
> a=fingerprint:SHA-256
> C1:E5:39:6E:FB:7A:97:5D:70:CF:65:EF:E7:5C:4D:37:1E:AE:9B:72:70:E4:C1:F4:
> 86:F8:7B:1A:8D:DE:B3:47
> a=sendrecv
> m=text 12836 RTP/SAVPF 105 106
> a=ice-ufrag:1d890d34098281a73af392833cdf7626
> a=ice-pwd:06e51e922264e51c5cc0209e3a1ec6a7
> a=candidate:Hc0a8026a 1 UDP 2130706431 192.168.2.106 12836 typ host
> a=candidate:S57a9bd66 1 UDP 1694498815 87.169.189.102 12836 typ srflx
> raddr 192.168.2.106 rport 12836 a=candidate:Hc0a8026a 2 UDP 2130706430
> 192.168.2.106 12837 typ host
> a=candidate:S57a9bd66 2 UDP 1694498814 87.169.189.102 12837 typ srflx
> raddr 192.168.2.106 rport 12837 a=connection:new a=setup:actpass
> a=fingerprint:SHA-256
> C1:E5:39:6E:FB:7A:97:5D:70:CF:65:EF:E7:5C:4D:37:1E:AE:9B:72:70:E4:C1:F4:
> 86:F8:7B:1A:8D:DE:B3:47
> a=rtpmap:105 RED/1000
> a=fmtp:105 106/106/106
> a=rtpmap:106 T140/1000
> a=sendrecv
>
> ---
>      -- Called SIP/6000
>
> <--- SIP read from WS:192.168.2.103:49848 ---> SIP/2.0 100 Trying
> Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
> To: <sip:bbglnljp at 72rvpk435t95.invalid;transport=ws>
> From: "6001" <sip:6001 at 192.168.2.106>;tag=as0ba3cd59
> Call-ID: 5681400a771147ed0c16fff2363c7e55 at 192.168.2.106:5060
> CSeq: 102 INVITE
> Supported: ice,replaces,outbound
> Content-Length: 0
>
> <------------->
> --- (8 headers 0 lines) ---
>
> <--- SIP read from WS:192.168.2.103:49848 ---> SIP/2.0 180 Ringing
> Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
> To: <sip:bbglnljp at 72rvpk435t95.invalid;transport=ws>;tag=tlkjkk4vl8
> From: "6001" <sip:6001 at 192.168.2.106>;tag=as0ba3cd59
> Call-ID: 5681400a771147ed0c16fff2363c7e55 at 192.168.2.106:5060
> CSeq: 102 INVITE
> Contact: <sip:bbglnljp at 72rvpk435t95.invalid;transport=ws>
> Supported: ice,replaces,outbound
> Content-Length: 0
>
> <------------->
> --- (9 headers 0 lines) ---
> list_route: route/path hop:
> <sip:bbglnljp at 72rvpk435t95.invalid;transport=ws>
>      -- SIP/6000-00000001 is ringing
>
> <--- Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 180 Ringing
> Via: SIP/2.0/WS
> 0iemcrsq9tm0.invalid;branch=z9hG4bK7826783;received=192.168.2.103;rport=
> 49851
> From: "6001" <sip:6001 at 192.168.2.106>;tag=m6bqn333dr
> To: <sip:6000 at 192.168.2.106>;tag=as1792125e
> Call-ID: 1ansppdrpdulbtr3j5ub
> CSeq: 6408 INVITE
> Server: Asterisk PBX 12.8.2
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:6000 at 192.168.2.106:5060;transport=WS>
> Content-Length: 0
>
>
> <------------>
>
> <--- SIP read from WS:192.168.2.103:49848 ---> SIP/2.0 488 Not
> Acceptable Here
> Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
> To: <sip:bbglnljp at 72rvpk435t95.invalid;transport=ws>;tag=tlkjkk4vl8
> From: "6001" <sip:6001 at 192.168.2.106>;tag=as0ba3cd59
> Call-ID: 5681400a771147ed0c16fff2363c7e55 at 192.168.2.106:5060
> CSeq: 102 INVITE
> Supported: ice,replaces,outbound
> Content-Length: 0
>
> <------------->
> --- (8 headers 0 lines) ---
> Transmitting (NAT) to 192.168.2.103:49848:
> ACK sip:bbglnljp at 72rvpk435t95.invalid;transport=ws SIP/2.0
> Via: SIP/2.0/WS 192.168.2.106:5060;branch=z9hG4bK53a0c8e1;rport
> Max-Forwards: 70
> From: "6001" <sip:6001 at 192.168.2.106>;tag=as0ba3cd59
> To: <sip:bbglnljp at 72rvpk435t95.invalid;transport=ws>;tag=tlkjkk4vl8
> Contact: <sip:6001 at 192.168.2.106:5060;transport=WS>
> Call-ID: 5681400a771147ed0c16fff2363c7e55 at 192.168.2.106:5060
> CSeq: 102 ACK
> User-Agent: Asterisk PBX 12.8.2
> Content-Length: 0
>
>
> ---
> Scheduling destruction of SIP dialog
> '5681400a771147ed0c16fff2363c7e55 at 192.168.2.106:5060' in 32000 ms
> (Method: INVITE)
>    == Everyone is busy/congested at this time (1:0/0/1)
>      -- Executing [6000 at outgoing:2] Answer("SIP/6001-00000000", "") in
> new stack Audio is at 19538 Adding codec 100003 (ulaw) to SDP Adding
> codec 100004 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding
> codec 100030 (opus) to SDP
>
> <--- Reliably Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 200
> OK
> Via: SIP/2.0/WS
> 0iemcrsq9tm0.invalid;branch=z9hG4bK7826783;received=192.168.2.103;rport=
> 49851
> From: "6001" <sip:6001 at 192.168.2.106>;tag=m6bqn333dr
> To: <sip:6000 at 192.168.2.106>;tag=as1792125e
> Call-ID: 1ansppdrpdulbtr3j5ub
> CSeq: 6408 INVITE
> Server: Asterisk PBX 12.8.2
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:6000 at 192.168.2.106:5060;transport=WS>
> Content-Type: application/sdp
> Content-Length: 1055
>
> v=0
> o=root 1700582523 1700582523 IN IP4 192.168.2.106 s=Asterisk PBX 12.8.2
> c=IN IP4 192.168.2.106 t=0 0 m=audio 19538 RTP/SAVPF 0 8 9 109
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:109 opus/48000/2
> a=fmtp:109
> maxplaybackrate=48000;sprop-
> maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-
> stereo=0;cbr=0;useinbandfec=0;usedtx=0
> a=ptime:20
> a=maxptime:60
> a=ice-ufrag:7fbfa28012692271620bb8c22da32ff3
> a=ice-pwd:30067a57115528082e8744df31454da4
> a=candidate:Hc0a8026a 1 UDP 2130706431 192.168.2.106 19538 typ host
> a=candidate:S57a9bd66 1 UDP 1694498815 87.169.189.102 19538 typ srflx
> raddr 192.168.2.106 rport 19538 a=candidate:Hc0a8026a 2 UDP 2130706430
> 192.168.2.106 19539 typ host
> a=candidate:S57a9bd66 2 UDP 1694498814 87.169.189.102 19539 typ srflx
> raddr 192.168.2.106 rport 19539 a=connection:new a=setup:active
> a=fingerprint:SHA-256
> C1:E5:39:6E:FB:7A:97:5D:70:CF:65:EF:E7:5C:4D:37:1E:AE:9B:72:70:E4:C1:F4:
> 86:F8:7B:1A:8D:DE:B3:47
> a=sendrecv
> m=video 0 UDP/TLS/RTP/SAVPF 120 126 97
>
> <------------>
>
> <--- SIP read from WS:192.168.2.103:49851 ---> ACK
> sip:6000 at 192.168.2.106:5060;transport=ws SIP/2.0
> Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK690056
> Max-Forwards: 69
> To: <sip:6000 at 192.168.2.106>;tag=as1792125e
> From: "6001" <sip:6001 at 192.168.2.106>;tag=m6bqn333dr
> Call-ID: 1ansppdrpdulbtr3j5ub
> CSeq: 6408 ACK
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
> Supported: outbound
> User-Agent: JsSIP 2.0.2
> Content-Length: 0
>
> <------------->
> --- (11 headers 0 lines) ---
>
> <--- SIP read from WS:192.168.2.103:49851 ---> BYE
> sip:6000 at 192.168.2.106:5060;transport=ws SIP/2.0
> Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK1426296
> Max-Forwards: 69
> To: <sip:6000 at 192.168.2.106>;tag=as1792125e
> From: "6001" <sip:6001 at 192.168.2.106>;tag=m6bqn333dr
> Call-ID: 1ansppdrpdulbtr3j5ub
> CSeq: 6409 BYE
> Reason: SIP ;cause=488; text="Not Acceptable Here"
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
> Supported: outbound
> User-Agent: JsSIP 2.0.2
> Content-Length: 0
>
> <------------->
> --- (12 headers 0 lines) ---
> Scheduling destruction of SIP dialog '1ansppdrpdulbtr3j5ub' in 32000 ms
> (Method: BYE)
>
> <--- Transmitting (NAT) to 192.168.2.103:49851 ---> SIP/2.0 200 OK
> Via: SIP/2.0/WS
> 0iemcrsq9tm0.invalid;branch=z9hG4bK1426296;received=192.168.2.103;rport=
> 49851
> From: "6001" <sip:6001 at 192.168.2.106>;tag=m6bqn333dr
> To: <sip:6000 at 192.168.2.106>;tag=as1792125e
> Call-ID: 1ansppdrpdulbtr3j5ub
> CSeq: 6409 BYE
> Server: Asterisk PBX 12.8.2
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
>
> May you please help me to make it word? i'm just interessting for the
> audio.
> Thank you in advance
>

Hi Olivier,

I am not sure what is the issue with your current setup, however I would 
like to point you to the latest Asterisk releases, i.e. >13.9.1. I think 
those releases include some fixes for issues related to WebRTC. For me 
theses releases work very well with WebRTC (audio & video).

There is only one remark I have to make: When using PJSIP I experienced 
an issue with the maximum size for SIP messages. I therefore had to 
manually increase it to be able to get it working.

Hope it helps,

Simon