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@192.168.2.106 SIP/2.0 Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK8394689 Max-Forwards: 69 To: <sip:6000@192.168.2.106> From: "6001" <sip:6001@192.168.2.106>;tag=m6bqn333dr Call-ID: 1ansppdrpdulbtr3j5ub CSeq: 6407 INVITE X-Can-Renegotiate: false Contact: <sip:5krseuop@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@192.168.2.106>;tag=m6bqn333dr To: <sip:6000@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@192.168.2.106 SIP/2.0 Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK8394689 To: <sip:6000@192.168.2.106>;tag=as28bc71f3 From: "6001" <sip:6001@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@192.168.2.106 SIP/2.0 Via: SIP/2.0/WS 0iemcrsq9tm0.invalid;branch=z9hG4bK7826783 Max-Forwards: 69 To: <sip:6000@192.168.2.106> From: "6001" <sip:6001@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@192.168.2.106", response="845417814e71ce56e93d846538ea31ae" X-Can-Renegotiate: false Contact: <sip:5krseuop@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@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@192.168.2.106>;tag=m6bqn333dr To: <sip:6000@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@192.168.2.106:5060;transport=WS> Content-Length: 0 <------------> -- Executing [6000@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@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@192.168.2.106>;tag=as0ba3cd59 To: <sip:bbglnljp@72rvpk435t95.invalid;transport=ws> Contact: <sip:6001@192.168.2.106:5060;transport=WS> Call-ID: 5681400a771147ed0c16fff2363c7e55@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@72rvpk435t95.invalid;transport=ws> From: "6001" <sip:6001@192.168.2.106>;tag=as0ba3cd59 Call-ID: 5681400a771147ed0c16fff2363c7e55@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@72rvpk435t95.invalid;transport=ws>;tag=tlkjkk4vl8 From: "6001" <sip:6001@192.168.2.106>;tag=as0ba3cd59 Call-ID: 5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060 CSeq: 102 INVITE Contact: <sip:bbglnljp@72rvpk435t95.invalid;transport=ws> Supported: ice,replaces,outbound Content-Length: 0 <-------------> --- (9 headers 0 lines) --- list_route: route/path hop: <sip:bbglnljp@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@192.168.2.106>;tag=m6bqn333dr To: <sip:6000@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@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@72rvpk435t95.invalid;transport=ws>;tag=tlkjkk4vl8 From: "6001" <sip:6001@192.168.2.106>;tag=as0ba3cd59 Call-ID: 5681400a771147ed0c16fff2363c7e55@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@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@192.168.2.106>;tag=as0ba3cd59 To: <sip:bbglnljp@72rvpk435t95.invalid;transport=ws>;tag=tlkjkk4vl8 Contact: <sip:6001@192.168.2.106:5060;transport=WS> Call-ID: 5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 12.8.2 Content-Length: 0 --- Scheduling destruction of SIP dialog '5681400a771147ed0c16fff2363c7e55@192.168.2.106:5060' in 32000 ms (Method: INVITE) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [6000@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@192.168.2.106>;tag=m6bqn333dr To: <sip:6000@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@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@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@192.168.2.106>;tag=as1792125e From: "6001" <sip:6001@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@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@192.168.2.106>;tag=as1792125e From: "6001" <sip:6001@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@192.168.2.106>;tag=m6bqn333dr To: <sip:6000@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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
