Hi,
First I would like to thank everyone for posting so much helpful material here. I have found this forum very valuable in fixing a lot of errors/ mistakes that were committed right from installation.
I am new to asterisk and have been experimenting with the idea of setting up WebRTC over SIP. I have succeeded in placing calls on plain and secure channels using Zoiper and Blink softphones respectively.
However I have encountered an error when attempting to use WebRTC to place calls. When the caller calls the callee, the callee’s phone rings and the caller’s side shows the callee’s phone ringing(I can see ‘SIP/2.0 180 Ringing’ on asterisk as well), but the call is immediately terminated as soon as the the call is picked up. This happens when I use SIP.js and JsSIP libraries for peers.
The client shows its status as ‘Terminated’. In the log asterisk shows both channels have joined a simple bridge but then leave immediately sending ‘BYE’. I have attached logs from both client phones as well asterisk’s own log along with the settings I am using. I had earlier used asterisk 13.1 but thinking it was a bug with the build I began to use asterisk 13.4 which is the version I have right now on my Linux box.
Environment:
OS : CentOS 6.6
Server : asterisk 13.4
Phones : JsSIP 0.6.34
Browser : Chrome 43.0.2357.134
I have googled and found many related problems but not this one. I have tried various NAT settings in “sip.conf” and even on different machines to diagnose the problem, but to no avail. In the beginning I had assumed this was a problem related to the IP seen in the SDP(It was showing our public IP) but the problem persisted even after setting ‘media_address’ to my local IP in"sip.conf" file. The entire setup is on our local LAN and hence I am inclined to rule out NATting problems
I have read and re-read several posts from this and other fora on peer settings that must be used to suit WebRTC. However, none of the changes in these settings were helpful in resolving the problem. I have now reverted to my original settings for these peers(named ‘2001’ and ‘2002’).
Any help in diagnosing and resolving this problem will be greatly appreciated. Apologies if the log is too verbose, but I thought more verbosity will help.
Here is the asterisk log(I had to truncate it because of a limit imposed by the website). As you can see in the end the two channels join the bridge and then puzzlingly leave it immediately. This did not happen in any of the non-WebRTC calls I had made before.
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from WS:192.168.1.42:53162 --->
INVITE sip:2002@192.168.1.111 SIP/2.0
Via: SIP/2.0/WS bon65t012pe4.invalid;branch=z9hG4bK3066115
Max-Forwards: 69
To: <sip:2002@192.168.1.111>
From: "2001" <sip:2001@192.168.1.111>;tag=igshfgl204
Call-ID: gb79ai0ntucb5qerigav
CSeq: 8600 INVITE
Authorization: Digest algorithm=MD5, username="2001", realm="192.168.1.111", nonce="7a8a6aa3", uri="sip:2002@192.168.1.111", response="ff3c3815292acb527800992cc86b305e"
X-Can-Renegotiate: true
Contact: <sip:pg9bv8gq@bon65t012pe4.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.34
Content-Length: 2945
v=0
o=- 4107294438299298130 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS hTRgdgqEGcIXe4XMuKSNLxSJjW2JoKY7Uu6s
m=audio 56458 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 202.65.138.134
a=rtcp:56459 IN IP4 202.65.138.134
a=candidate:2447636755 1 udp 2122260223 192.168.1.42 56458 typ host generation 0
a=candidate:2447636755 2 udp 2122260222 192.168.1.42 56459 typ host generation 0
a=candidate:1744011463 1 udp 1686052607 202.65.138.134 56458 typ srflx raddr 192.168.1.42 rport 56458 generation 0
a=candidate:1744011463 2 udp 1686052606 202.65.138.134 56459 typ srflx raddr 192.168.1.42 rport 56459 generation 0
a=candidate:3747612131 1 tcp 1518280447 192.168.1.42 0 typ host tcptype active generation 0
a=candidate:3747612131 2 tcp 1518280446 192.168.1.42 0 typ host tcptype active generation 0
a=ice-ufrag:nUU6+JG5ApEwV8+Z
a=ice-pwd:elKN+JOdyo74RaCGY+xZm/pq
a=fingerprint:sha-256 9A:45:89:15:3A:3D:A1:E6:99:B1:49:D2:91:05:10:84:76:59:54:57:1A:2C:64:EC:A4:E6:00:3B:B7:65:99:58
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:929327403 cname:7U/toGGAtQ+eewFx
a=ssrc:929327403 msid:hTRgdgqEGcIXe4XMuKSNLxSJjW2JoKY7Uu6s 1e8340cf-dfaf-4611-85cd-324e4bab8f4b
a=ssrc:929327403 mslabel:hTRgdgqEGcIXe4XMuKSNLxSJjW2JoKY7Uu6s
a=ssrc:929327403 label:1e8340cf-dfaf-4611-85cd-324e4bab8f4b
m=video 1030 RTP/SAVPF 100 116 117 96
c=IN IP4 202.65.138.134
a=rtcp:56461 IN IP4 192.168.1.42
a=candidate:2447636755 1 udp 2122260223 192.168.1.42 56460 typ host generation 0
a=candidate:2447636755 2 udp 2122260222 192.168.1.42 56461 typ host generation 0
a=candidate:1744011463 1 udp 1686052607 202.65.138.134 1030 typ srflx raddr 192.168.1.42 rport 56460 generation 0
a=candidate:3747612131 1 tcp 1518280447 192.168.1.42 0 typ host tcptype active generation 0
a=candidate:3747612131 2 tcp 1518280446 192.168.1.42 0 typ host tcptype active generation 0
a=ice-ufrag:nUU6+JG5ApEwV8+Z
a=ice-pwd:elKN+JOdyo74RaCGY+xZm/pq
a=fingerprint:sha-256 9A:45:89:15:3A:3D:A1:E6:99:B1:49:D2:91:05:10:84:76:59:54:57:1A:2C:64:EC:A4:E6:00:3B:B7:65:99:58
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=recvonly
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
<------------->
--- (16 headers 67 lines) ---
Using INVITE request as basis request - gb79ai0ntucb5qerigav
Found peer '2001' for '2001' from 192.168.1.42:53162
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Found RTP audio format 111
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format ISAC for ID 103
Found unknown media description format ISAC for ID 104
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 106
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found audio description format telephone-event for ID 126
Found RTP video format 100
Found RTP video format 116
Found RTP video format 117
Found RTP video format 96
Found video description format VP8 for ID 100
failed to extend from 64 to 98
Capabilities: us - (g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|), peer - audio=(ulaw|alaw|g722|opus)/video=(vp8|g719|speex16)/text=(nothing), combined - (ulaw|alaw|g722|opus|vp8)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 202.65.138.134:56458
Peer video RTP is at port 202.65.138.134:1030
Peer doesn't provide T.140
Looking for 2002 in WebRTCContext (domain 192.168.1.111)
failed to extend from 64 to 98
sip_route_dump: route/path hop: <sip:pg9bv8gq@bon65t012pe4.invalid;transport=ws;ob>
<--- Transmitting (NAT) to 192.168.1.42:53162 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS bon65t012pe4.invalid;branch=z9hG4bK3066115;received=192.168.1.42;rport=53162
From: "2001" <sip:2001@192.168.1.111>;tag=igshfgl204
To: <sip:2002@192.168.1.111>
Call-ID: gb79ai0ntucb5qerigav
CSeq: 8600 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:2002@192.168.1.111:5060;transport=WS>
Content-Length: 0
<------------>
-- Executing [2002@WebRTCContext:1] Dial("SIP/2001-00000002", "SIP/2002,20") in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
failed to extend from 64 to 98
We think we can do text
And we have a text rtp object
failed to extend from 64 to 97
Audio is at 13114
Video is at 192.168.1.111:16242
Lets set up the text sdp
Text is at 0.0.0.0:19848
Adding codec ulaw to SDP
Adding video codec vp8 to SDP
Adding codec g723 to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding video codec h261 to SDP
Adding video codec h263 to SDP
Adding video codec h263p to SDP
Adding video codec h264 to SDP
Adding video codec mpeg4 to SDP
Adding text codec red to SDP
Adding text codec t140 to SDP
Adding codec none to SDP
Adding non-codec 0x1 (telephone-event) to SDP
failed to extend from 64 to 97
Reliably Transmitting (NAT) to 192.168.1.207:31060:
INVITE sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.111:5060;branch=z9hG4bK3101cfad;rport
Max-Forwards: 70
From: "2001" <sip:2001@192.168.1.111>;tag=as6202d472
To: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws>
Contact: <sip:2001@192.168.1.111:5060;transport=WS>
Call-ID: 6fcdb89144bbcfbd31290a0465f62e21@192.168.1.111:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.4.0
Date: Tue, 28 Jul 2015 04:21:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 3436
v=0
o=root 1453506867 1453506867 IN IP4 192.168.1.111
s=Asterisk PBX 13.4.0
c=IN IP4 192.168.1.111
b=CT:384
t=0 0
m=audio 13114 RTP/SAVPF 0 4 8 3 111 112 5 10 118 7 18 110 117 119 97 9 102 115 116 107 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
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:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
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:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=ice-ufrag:0c22106840c314b809ff69f2260f801f
a=ice-pwd:57a7a15115a0bf246865892416b92aea
a=candidate:Hc0a8016f 1 UDP 2130706431 192.168.1.111 13114 typ host
a=candidate:Hc0a87a01 1 UDP 2130706431 192.168.122.1 13114 typ host
a=candidate:Sca418a86 1 UDP 1694498815 202.65.138.134 13114 typ srflx raddr 192.168.1.111 rport 13114
a=candidate:Hc0a8016f 2 UDP 2130706430 192.168.1.111 13115 typ host
a=candidate:Hc0a87a01 2 UDP 2130706430 192.168.122.1 13115 typ host
a=candidate:Sca418a86 2 UDP 1694498814 202.65.138.134 13115 typ srflx raddr 192.168.1.111 rport 13115
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 F9:BA:AC:66:E8:B0:8F:A0:3B:3A:8F:66:6D:3C:2B:E1:0A:F7:74:E5:C2:01:01:41:59:21:6B:27:72:F5:21:32
a=sendrecv
m=video 16242 RTP/SAVPF 100 31 34 98 99 104
a=ice-ufrag:58a215e23ece559960f0945934ea95b5
a=ice-pwd:58a5bf3b3f7c26aa24bc4a7a12a4c628
a=candidate:Hc0a8016f 1 UDP 2130706431 192.168.1.111 16242 typ host
a=candidate:Hc0a87a01 1 UDP 2130706431 192.168.122.1 16242 typ host
a=candidate:Sca418a86 1 UDP 1694498815 202.65.138.134 16242 typ srflx raddr 192.168.1.111 rport 16242
a=candidate:Hc0a8016f 2 UDP 2130706430 192.168.1.111 16243 typ host
a=candidate:Hc0a87a01 2 UDP 2130706430 192.168.122.1 16243 typ host
a=candidate:Sca418a86 2 UDP 1694498814 202.65.138.134 16243 typ srflx raddr 192.168.1.111 rport 16243
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 F9:BA:AC:66:E8:B0:8F:A0:3B:3A:8F:66:6D:3C:2B:E1:0A:F7:74:E5:C2:01:01:41:59:21:6B:27:72:F5:21:32
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=sendrecv
m=text 19848 RTP/SAVPF 105 106
a=ice-ufrag:7fd8b87a1c2af1d81a70cc7424c4bb48
a=ice-pwd:21b7205030e101b13d1e44b164a32588
a=candidate:Hc0a8016f 1 UDP 2130706431 192.168.1.111 19848 typ host
a=candidate:Hc0a87a01 1 UDP 2130706431 192.168.122.1 19848 typ host
a=candidate:Sca418a86 1 UDP 1694498815 202.65.138.134 19848 typ srflx raddr 192.168.1.111 rport 19848
a=candidate:Hc0a8016f 2 UDP 2130706430 192.168.1.111 19849 typ host
a=candidate:Hc0a87a01 2 UDP 2130706430 192.168.122.1 19849 typ host
a=candidate:Sca418a86 2 UDP 1694498814 202.65.138.134 19849 typ srflx raddr 192.168.1.111 rport 19849
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 F9:BA:AC:66:E8:B0:8F:A0:3B:3A:8F:66:6D:3C:2B:E1:0A:F7:74:E5:C2:01:01:41:59:21:6B:27:72:F5:21:32
a=rtpmap:105 RED/1000
a=fmtp:105 106/106/106
a=rtpmap:106 T140/1000
a=sendrecv
---
-- Called SIP/2002
<--- SIP read from WS:192.168.1.207:31060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 192.168.1.111:5060;branch=z9hG4bK3101cfad;rport
To: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws>
From: "2001" <sip:2001@192.168.1.111>;tag=as6202d472
Call-ID: 6fcdb89144bbcfbd31290a0465f62e21@192.168.1.111:5060
CSeq: 102 INVITE
Supported: timer,ice,outbound
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from WS:192.168.1.207:31060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.1.111:5060;branch=z9hG4bK3101cfad;rport
To: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws>;tag=sah8g26qot
From: "2001" <sip:2001@192.168.1.111>;tag=as6202d472
Call-ID: 6fcdb89144bbcfbd31290a0465f62e21@192.168.1.111:5060
CSeq: 102 INVITE
Contact: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws>
Supported: timer,ice,outbound
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws>
-- SIP/2002-00000003 is ringing
<--- Transmitting (NAT) to 192.168.1.42:53162 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS bon65t012pe4.invalid;branch=z9hG4bK3066115;received=192.168.1.42;rport=53162
From: "2001" <sip:2001@192.168.1.111>;tag=igshfgl204
To: <sip:2002@192.168.1.111>;tag=as1b79aab8
Call-ID: gb79ai0ntucb5qerigav
CSeq: 8600 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:2002@192.168.1.111:5060;transport=WS>
Content-Length: 0
<------------>
<--- SIP read from WS:192.168.1.207:31060 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.1.111:5060;branch=z9hG4bK3101cfad;rport
To: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws>;tag=sah8g26qot
From: "2001" <sip:2001@192.168.1.111>;tag=as6202d472
Call-ID: 6fcdb89144bbcfbd31290a0465f62e21@192.168.1.111:5060
CSeq: 102 INVITE
Contact: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws>
X-Can-Renegotiate: true
Session-Expires: 90;refresher=uas
Supported: timer,ice,outbound
Content-Type: application/sdp
Content-Length: 2696
v=0
o=- 7485707620799868540 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS h3wh3zAjBe8Yg3loXLWxPwu7QaG2p93rX4K0
m=audio 52571 RTP/SAVPF 0 8 9 107 101
c=IN IP4 202.65.138.134
a=rtcp:52572 IN IP4 202.65.138.134
a=candidate:2833646159 1 udp 2122260223 192.168.1.207 52571 typ host generation 0
a=candidate:2833646159 2 udp 2122260222 192.168.1.207 52572 typ host generation 0
a=candidate:699270395 2 udp 1686052606 202.65.138.134 52572 typ srflx raddr 192.168.1.207 rport 52572 generation 0
a=candidate:699270395 1 udp 1686052607 202.65.138.134 52571 typ srflx raddr 192.168.1.207 rport 52571 generation 0
a=candidate:3865444031 1 tcp 1518280447 192.168.1.207 0 typ host tcptype active generation 0
a=candidate:3865444031 2 tcp 1518280446 192.168.1.207 0 typ host tcptype active generation 0
a=ice-ufrag:jw7On0JYhfcueEc3
a=ice-pwd:yR7QU2CJqA4jJ9kO9pV9E4Dy
a=fingerprint:sha-256 76:B5:A9:FF:2C:2E:15:EB:D2:EE:18:4D:B3:23:DD:17:CA:48:B4:68:31:E6:A0:72:C5:69:E8:E7:0E:FC:D9:29
a=setup:active
a=mid:audio
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:107 opus/48000/2
a=fmtp:107 minptime=10; useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=maxptime:60
a=ssrc:4208219454 cname:6a7Kb3MjG+RwOVkj
a=ssrc:4208219454 msid:h3wh3zAjBe8Yg3loXLWxPwu7QaG2p93rX4K0 e0e18f08-3a9b-4508-87ca-08864b31ecbe
a=ssrc:4208219454 mslabel:h3wh3zAjBe8Yg3loXLWxPwu7QaG2p93rX4K0
a=ssrc:4208219454 label:e0e18f08-3a9b-4508-87ca-08864b31ecbe
m=video 52573 RTP/SAVPF 100
c=IN IP4 202.65.138.134
a=rtcp:52574 IN IP4 202.65.138.134
a=candidate:2833646159 1 udp 2122260223 192.168.1.207 52573 typ host generation 0
a=candidate:2833646159 2 udp 2122260222 192.168.1.207 52574 typ host generation 0
a=candidate:699270395 1 udp 1686052607 202.65.138.134 52573 typ srflx raddr 192.168.1.207 rport 52573 generation 0
a=candidate:699270395 2 udp 1686052606 202.65.138.134 52574 typ srflx raddr 192.168.1.207 rport 52574 generation 0
a=candidate:3865444031 1 tcp 1518280447 192.168.1.207 0 typ host tcptype active generation 0
a=candidate:3865444031 2 tcp 1518280446 192.168.1.207 0 typ host tcptype active generation 0
a=ice-ufrag:GzMjZIGht6jj/wkG
a=ice-pwd:s1rrNhl4WCFYbncv5EnQz2d3
a=fingerprint:sha-256 76:B5:A9:FF:2C:2E:15:EB:D2:EE:18:4D:B3:23:DD:17:CA:48:B4:68:31:E6:A0:72:C5:69:E8:E7:0E:FC:D9:29
a=setup:active
a=mid:video
a=sendrecv
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=ssrc:1699206159 cname:6a7Kb3MjG+RwOVkj
a=ssrc:1699206159 msid:h3wh3zAjBe8Yg3loXLWxPwu7QaG2p93rX4K0 424ad54b-cfc0-4cdf-b199-d690fd6f5b8e
a=ssrc:1699206159 mslabel:h3wh3zAjBe8Yg3loXLWxPwu7QaG2p93rX4K0
a=ssrc:1699206159 label:424ad54b-cfc0-4cdf-b199-d690fd6f5b8e
<------------->
--- (12 headers 52 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 107
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format opus for ID 107
Found audio description format telephone-event for ID 101
Found RTP video format 100
Found video description format VP8 for ID 100
failed to extend from 64 to 98
Capabilities: us - (g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|), peer - audio=(ulaw|alaw|g722|opus)/video=(vp8)/text=(nothing), combined - (ulaw|alaw|g722|opus|vp8)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 202.65.138.134:52571
Peer video RTP is at port 202.65.138.134:52573
Peer doesn't provide T.140
sip_route_dump: route/path hop: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws>
Transmitting (NAT) to 192.168.1.207:31060:
ACK sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.111:5060;branch=z9hG4bK1d8b005f;rport
Max-Forwards: 70
From: "2001" <sip:2001@192.168.1.111>;tag=as6202d472
To: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws>;tag=sah8g26qot
Contact: <sip:2001@192.168.1.111:5060;transport=WS>
Call-ID: 6fcdb89144bbcfbd31290a0465f62e21@192.168.1.111:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.4.0
Content-Length: 0
---
-- SIP/2002-00000003 answered SIP/2001-00000002
Audio is at 13340
Video is at 192.168.1.111:11220
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec g722 to SDP
Adding codec opus to SDP
Adding video codec vp8 to SDP
Adding codec g723 to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec g729 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding video codec h261 to SDP
Adding video codec h263 to SDP
Adding video codec h263p to SDP
Adding video codec h264 to SDP
Adding video codec mpeg4 to SDP
Adding codec none to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 192.168.1.42:53162 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS bon65t012pe4.invalid;branch=z9hG4bK3066115;received=192.168.1.42;rport=53162
From: "2001" <sip:2001@192.168.1.111>;tag=igshfgl204
To: <sip:2002@192.168.1.111>;tag=as1b79aab8
Call-ID: gb79ai0ntucb5qerigav
CSeq: 8600 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:2002@192.168.1.111:5060;transport=WS>
Content-Type: application/sdp
Require: timer
Content-Length: 2742
v=0
o=root 651895664 651895664 IN IP4 192.168.1.111
s=Asterisk PBX 13.4.0
c=IN IP4 192.168.1.111
b=CT:384
t=0 0
m=audio 13340 RTP/SAVPF 0 8 9 111 4 3 111 112 5 10 118 7 18 110 117 119 97 102 115 116 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
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:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
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:126 telephone-event/8000
a=fmtp:126 0-16
a=maxptime:20
a=ice-ufrag:3ab36cb829077d8442787a0a5b1f2749
a=ice-pwd:0cea67bf489fdaa03334b0cb58befc4e
a=candidate:Hc0a8016f 1 UDP 2130706431 192.168.1.111 13340 typ host
a=candidate:Hc0a87a01 1 UDP 2130706431 192.168.122.1 13340 typ host
a=candidate:Sca418a86 1 UDP 1694498815 202.65.138.134 13340 typ srflx raddr 192.168.1.111 rport 13340
a=candidate:Hc0a8016f 2 UDP 2130706430 192.168.1.111 13341 typ host
a=candidate:Hc0a87a01 2 UDP 2130706430 192.168.122.1 13341 typ host
a=candidate:Sca418a86 2 UDP 1694498814 202.65.138.134 13341 typ srflx raddr 192.168.1.111 rport 13341
a=connection:new
a=setup:active
a=fingerprint:SHA-256 F9:BA:AC:66:E8:B0:8F:A0:3B:3A:8F:66:6D:3C:2B:E1:0A:F7:74:E5:C2:01:01:41:59:21:6B:27:72:F5:21:32
a=sendrecv
m=video 11220 RTP/SAVPF 100 31 34 98 99 104
a=ice-ufrag:5fb6ecd97037f7376ae41a3b2a374213
a=ice-pwd:30eb051708aa4b095813fbed304beccd
a=candidate:Hc0a8016f 1 UDP 2130706431 192.168.1.111 11220 typ host
a=candidate:Hc0a87a01 1 UDP 2130706431 192.168.122.1 11220 typ host
a=candidate:Sca418a86 1 UDP 1694498815 202.65.138.134 11220 typ srflx raddr 192.168.1.111 rport 11220
a=candidate:Hc0a8016f 2 UDP 2130706430 192.168.1.111 11221 typ host
a=candidate:Hc0a87a01 2 UDP 2130706430 192.168.122.1 11221 typ host
a=candidate:Sca418a86 2 UDP 1694498814 202.65.138.134 11221 typ srflx raddr 192.168.1.111 rport 11221
a=connection:new
a=setup:active
a=fingerprint:SHA-256 F9:BA:AC:66:E8:B0:8F:A0:3B:3A:8F:66:6D:3C:2B:E1:0A:F7:74:E5:C2:01:01:41:59:21:6B:27:72:F5:21:32
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=sendrecv
<------------>
-- Channel SIP/2002-00000003 joined 'simple_bridge' basic-bridge <e9581d10-38e7-4865-bbe8-df9bf4461549>
-- Channel SIP/2001-00000002 joined 'simple_bridge' basic-bridge <e9581d10-38e7-4865-bbe8-df9bf4461549>
-- Channel SIP/2002-00000003 left 'simple_bridge' basic-bridge <e9581d10-38e7-4865-bbe8-df9bf4461549>
-- Channel SIP/2001-00000002 left 'simple_bridge' basic-bridge <e9581d10-38e7-4865-bbe8-df9bf4461549>
== Spawn extension (WebRTCContext, 2002, 1) exited non-zero on 'SIP/2001-00000002'
Scheduling destruction of SIP dialog 'gb79ai0ntucb5qerigav' in 32000 ms (Method: INVITE)
Scheduling destruction of SIP dialog '6fcdb89144bbcfbd31290a0465f62e21@192.168.1.111:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.1.207:31060:
BYE sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.1.111:5060;branch=z9hG4bK4e12df21;rport
Max-Forwards: 70
From: "2001" <sip:2001@192.168.1.111>;tag=as6202d472
To: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws>;tag=sah8g26qot
Call-ID: 6fcdb89144bbcfbd31290a0465f62e21@192.168.1.111:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.4.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
The log from peer 1:
jssip.js:21621 JsSIP:RTCSession new +1ms
jssip.js:21621 JsSIP:RTCSession connect() +1ms
jssip.js:21621 rtcninja:RTCPeerConnection new | pcConfig: Object +5ms
jssip.js:21621 rtcninja:RTCPeerConnection setConfigurationAndOptions | processed pcConfig: Object +10ms
jssip.js:21621 JsSIP:RTCSession newRTCSession +5ms
gui.js:73 Tryit: getSession
gui.js:23 Tryit: createSession
gui.js:73 Tryit: getSession
gui.js:98 Tryit: createCompositionIndicator
iscomposing.js:1146 iscomposing:CompositionIndicator new() | [options:Object] +0ms
iscomposing.js:1146 iscomposing:Composer new() | processed options [format:0, refreshInterval:120, idleTimeout:15] +3ms
iscomposing.js:1146 iscomposing:Receiver new() | processed options [format:0] +3ms
iscomposing.js:1146 iscomposing:CompositionIndicator idle() +2ms
iscomposing.js:1146 iscomposing:Composer setStatus() | from IDLE to IDLE +1ms
gui.js:89 Tryit: renderSessions
sessions.jsx:279 Tryit: ChatBox::componentDidMount()
sessions.jsx:308 Tryit: ChatBox::handleFocus()
sessions.jsx:33 Tryit: Session::componentMount()
sessions.jsx:75 Tryit: Session::registerCallSession()
sessions.jsx:139 Tryit: Session::setCallStatus()
sessions.jsx:284 Tryit: ChatBox::componentDidUpdate()
sessions.jsx:46 Tryit: Session::componentDidUpdate()
jssip.js:21621 rtcninja:Adapter getUserMedia() | constraints: Object +55ms
sessions.jsx:302 Tryit: ChatBox::handleBlur()
gui.js:488 Tryit: chatInputBlur
gui.js:73 Tryit: getSession
iscomposing.js:1146 iscomposing:CompositionIndicator idle() +54ms
iscomposing.js:1146 iscomposing:Composer setStatus() | from IDLE to IDLE +1ms
sessions.jsx:308 Tryit: ChatBox::handleFocus()
jssip.js:21621 rtcninja:Adapter getUserMedia() | success +1s
jssip.js:21621 rtcninja:RTCPeerConnection addStream() | stream: MediaStream +1ms
jssip.js:21621 JsSIP:RTCSession session connecting +7ms
jssip.js:21621 rtcninja:RTCPeerConnection getLocalStreams() +1ms
jssip.js:21621 rtcninja:RTCPeerConnection getLocalStreams() +0ms
jssip.js:21621 JsSIP:RTCSession createLocalDescription() +1ms
jssip.js:21621 rtcninja:RTCPeerConnection createOffer() +2ms
jssip.js:21621 rtcninja:RTCPeerConnection createOffer() | success +2ms
jssip.js:21621 rtcninja:RTCPeerConnection setLocalDescription() +1ms
jssip.js:21621 rtcninja:RTCPeerConnection onnegotiationneeded() +1ms
jssip.js:21621 rtcninja:RTCPeerConnection onsignalingstatechange() | signalingState: have-local-offer +8ms
jssip.js:21621 rtcninja:RTCPeerConnection setLocalDescription() | success +23ms
jssip.js:21621 rtcninja:RTCPeerConnection setLocalDescription() | ending gathering in 2000 ms (gatheringTimeout option) +1ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:2833646159 1 udp 2122260223 192.168.1.207 53933 typ host generation 0 +2ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:2833646159 2 udp 2122260222 192.168.1.207 53934 typ host generation 0 +1ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m1(video) candidate:2833646159 1 udp 2122260223 192.168.1.207 53935 typ host generation 0 +0ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m1(video) candidate:2833646159 2 udp 2122260222 192.168.1.207 53936 typ host generation 0 +1ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m1(video) candidate:699270395 2 udp 1686052606 202.65.138.134 53936 typ srflx raddr 192.168.1.207 rport 53936 generation 0 +80ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m1(video) candidate:699270395 1 udp 1686052607 202.65.138.134 53935 typ srflx raddr 192.168.1.207 rport 53935 generation 0 +4ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:699270395 2 udp 1686052606 202.65.138.134 53934 typ srflx raddr 192.168.1.207 rport 53934 generation 0 +1ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:699270395 1 udp 1686052607 202.65.138.134 53933 typ srflx raddr 192.168.1.207 rport 53933 generation 0 +5ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:3865444031 1 tcp 1518280447 192.168.1.207 0 typ host tcptype active generation 0 +6ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m0(audio) candidate:3865444031 2 tcp 1518280446 192.168.1.207 0 typ host tcptype active generation 0 +0ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m1(video) candidate:3865444031 1 tcp 1518280447 192.168.1.207 0 typ host tcptype active generation 0 +1ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | m1(video) candidate:3865444031 2 tcp 1518280446 192.168.1.207 0 typ host tcptype active generation 0 +0ms
jssip.js:21621 rtcninja:RTCPeerConnection onicecandidate() | end of candidates +49ms
jssip.js:21621 JsSIP:Transport sending WebSocket message:
INVITE sip:2001@192.168.1.111 SIP/2.0
Via: SIP/2.0/WS sb9dhbvefrf4.invalid;branch=z9hG4bK7921786
Max-Forwards: 69
To: <sip:2001@192.168.1.111>
From: "2002" <sip:2002@192.168.1.111>;tag=omj9k66u8n
Call-ID: a5qndau4t06rbms2k65l
CSeq: 869 INVITE
X-Can-Renegotiate: true
Contact: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.34
Content-Length: 3077
v=0
o=- 6990479560176255313 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS 6TSOqyPl1SXJiCc7ggYiAgGrP2w6TQ4ZjbVI
m=audio 53933 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 202.65.138.134
a=rtcp:53934 IN IP4 202.65.138.134
a=candidate:2833646159 1 udp 2122260223 192.168.1.207 53933 typ host generation 0
a=candidate:2833646159 2 udp 2122260222 192.168.1.207 53934 typ host generation 0
a=candidate:699270395 2 udp 1686052606 202.65.138.134 53934 typ srflx raddr 192.168.1.207 rport 53934 generation 0
a=candidate:699270395 1 udp 1686052607 202.65.138.134 53933 typ srflx raddr 192.168.1.207 rport 53933 generation 0
a=candidate:3865444031 1 tcp 1518280447 192.168.1.207 0 typ host tcptype active generation 0
a=candidate:3865444031 2 tcp 1518280446 192.168.1.207 0 typ host tcptype active generation 0
a=ice-ufrag:KA24vzqgiDqNEB81
a=ice-pwd:KnmVF8JSy6Ya/zq8uhHzhkF2
a=fingerprint:sha-256 76:B5:A9:FF:2C:2E:15:EB:D2:EE:18:4D:B3:23:DD:17:CA:48:B4:68:31:E6:A0:72:C5:69:E8:E7:0E:FC:D9:29
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2197929744 cname:txo8vtmSa4SJYwiv
a=ssrc:2197929744 msid:6TSOqyPl1SXJiCc7ggYiAgGrP2w6TQ4ZjbVI a9fe75eb-2f88-4867-823d-014549ef041a
a=ssrc:2197929744 mslabel:6TSOqyPl1SXJiCc7ggYiAgGrP2w6TQ4ZjbVI
a=ssrc:2197929744 label:a9fe75eb-2f88-4867-823d-014549ef041a
m=video 53935 RTP/SAVPF 100 116 117 96
c=IN IP4 202.65.138.134
a=rtcp:53936 IN IP4 202.65.138.134
a=candidate:2833646159 1 udp 2122260223 192.168.1.207 53935 typ host generation 0
a=candidate:2833646159 2 udp 2122260222 192.168.1.207 53936 typ host generation 0
a=candidate:699270395 2 udp 1686052606 202.65.138.134 53936 typ srflx raddr 192.168.1.207 rport 53936 generation 0
a=candidate:699270395 1 udp 1686052607 202.65.138.134 53935 typ srflx raddr 192.168.1.207 rport 53935 generation 0
a=candidate:3865444031 1 tcp 1518280447 192.168.1.207 0 typ host tcptype active generation 0
a=candidate:3865444031 2 tcp 1518280446 192.168.1.207 0 typ host tcptype active generation 0
a=ice-ufrag:KA24vzqgiDqNEB81
a=ice-pwd:KnmVF8JSy6Ya/zq8uhHzhkF2
a=fingerprint:sha-256 76:B5:A9:FF:2C:2E:15:EB:D2:EE:18:4D:B3:23:DD:17:CA:48:B4:68:31:E6:A0:72:C5:69:E8:E7:0E:FC:D9:29
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=recvonly
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
+4ms
jssip.js:21621 JsSIP:Transport received WebSocket text message:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS sb9dhbvefrf4.invalid;branch=z9hG4bK7921786;received=192.168.1.207;rport=31060
From: "2002" <sip:2002@192.168.1.111>;tag=omj9k66u8n
To: <sip:2001@192.168.1.111>;tag=as53ea2312
Call-ID: a5qndau4t06rbms2k65l
CSeq: 869 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.111", nonce="014cb58e"
Content-Length: 0
+3ms
jssip.js:21621 JsSIP:Transport sending WebSocket message:
ACK sip:2001@192.168.1.111 SIP/2.0
Via: SIP/2.0/WS sb9dhbvefrf4.invalid;branch=z9hG4bK7921786
To: <sip:2001@192.168.1.111>;tag=as53ea2312
From: "2002" <sip:2002@192.168.1.111>;tag=omj9k66u8n
Call-ID: a5qndau4t06rbms2k65l
CSeq: 869 ACK
Content-Length: 0
+9ms
jssip.js:21621 JsSIP:Transport sending WebSocket message:
INVITE sip:2001@192.168.1.111 SIP/2.0
Via: SIP/2.0/WS sb9dhbvefrf4.invalid;branch=z9hG4bK6330132
Max-Forwards: 69
To: <sip:2001@192.168.1.111>
From: "2002" <sip:2002@192.168.1.111>;tag=omj9k66u8n
Call-ID: a5qndau4t06rbms2k65l
CSeq: 870 INVITE
Authorization: Digest algorithm=MD5, username="2002", realm="192.168.1.111", nonce="014cb58e", uri="sip:2001@192.168.1.111", response="ee66343c093d4e82682f5bf6b48a9e7d"
X-Can-Renegotiate: true
Contact: <sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.34
Content-Length: 3077
v=0
o=- 6990479560176255313 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS 6TSOqyPl1SXJiCc7ggYiAgGrP2w6TQ4ZjbVI
m=audio 53933 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 202.65.138.134
a=rtcp:53934 IN IP4 202.65.138.134
a=candidate:2833646159 1 udp 2122260223 192.168.1.207 53933 typ host generation 0
a=candidate:2833646159 2 udp 2122260222 192.168.1.207 53934 typ host generation 0
a=candidate:699270395 2 udp 1686052606 202.65.138.134 53934 typ srflx raddr 192.168.1.207 rport 53934 generation 0
a=candidate:699270395 1 udp 1686052607 202.65.138.134 53933 typ srflx raddr 192.168.1.207 rport 53933 generation 0
a=candidate:3865444031 1 tcp 1518280447 192.168.1.207 0 typ host tcptype active generation 0
a=candidate:3865444031 2 tcp 1518280446 192.168.1.207 0 typ host tcptype active generation 0
a=ice-ufrag:KA24vzqgiDqNEB81
a=ice-pwd:KnmVF8JSy6Ya/zq8uhHzhkF2
a=fingerprint:sha-256 76:B5:A9:FF:2C:2E:15:EB:D2:EE:18:4D:B3:23:DD:17:CA:48:B4:68:31:E6:A0:72:C5:69:E8:E7:0E:FC:D9:29
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:2197929744 cname:txo8vtmSa4SJYwiv
a=ssrc:2197929744 msid:6TSOqyPl1SXJiCc7ggYiAgGrP2w6TQ4ZjbVI a9fe75eb-2f88-4867-823d-014549ef041a
a=ssrc:2197929744 mslabel:6TSOqyPl1SXJiCc7ggYiAgGrP2w6TQ4ZjbVI
a=ssrc:2197929744 label:a9fe75eb-2f88-4867-823d-014549ef041a
m=video 53935 RTP/SAVPF 100 116 117 96
c=IN IP4 202.65.138.134
a=rtcp:53936 IN IP4 202.65.138.134
a=candidate:2833646159 1 udp 2122260223 192.168.1.207 53935 typ host generation 0
a=candidate:2833646159 2 udp 2122260222 192.168.1.207 53936 typ host generation 0
a=candidate:699270395 2 udp 1686052606 202.65.138.134 53936 typ srflx raddr 192.168.1.207 rport 53936 generation 0
a=candidate:699270395 1 udp 1686052607 202.65.138.134 53935 typ srflx raddr 192.168.1.207 rport 53935 generation 0
a=candidate:3865444031 1 tcp 1518280447 192.168.1.207 0 typ host tcptype active generation 0
a=candidate:3865444031 2 tcp 1518280446 192.168.1.207 0 typ host tcptype active generation 0
a=ice-ufrag:KA24vzqgiDqNEB81
a=ice-pwd:KnmVF8JSy6Ya/zq8uhHzhkF2
a=fingerprint:sha-256 76:B5:A9:FF:2C:2E:15:EB:D2:EE:18:4D:B3:23:DD:17:CA:48:B4:68:31:E6:A0:72:C5:69:E8:E7:0E:FC:D9:29
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:4 urn:3gpp:video-orientation
a=recvonly
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
+2ms
jssip.js:21621 JsSIP:InviteClientTransaction Timer D expired for transaction z9hG4bK7921786 +1ms
jssip.js:21621 JsSIP:Transport received WebSocket text message:
SIP/2.0 100 Trying
Via: SIP/2.0/WS sb9dhbvefrf4.invalid;branch=z9hG4bK6330132;received=192.168.1.207;rport=31060
From: "2002" <sip:2002@192.168.1.111>;tag=omj9k66u8n
To: <sip:2001@192.168.1.111>
Call-ID: a5qndau4t06rbms2k65l
CSeq: 870 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:2001@192.168.1.111:5060;transport=WS>
Content-Length: 0
+519ms
jssip.js:21621 JsSIP:RTCSession receiveInviteResponse() +10ms
jssip.js:21621 JsSIP:Transport received WebSocket text message:
SIP/2.0 180 Ringing
Via: SIP/2.0/WS sb9dhbvefrf4.invalid;branch=z9hG4bK6330132;received=192.168.1.207;rport=31060
From: "2002" <sip:2002@192.168.1.111>;tag=omj9k66u8n
To: <sip:2001@192.168.1.111>;tag=as3868ee20
Call-ID: a5qndau4t06rbms2k65l
CSeq: 870 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:2001@192.168.1.111:5060;transport=WS>
Content-Length: 0
+589ms
jssip.js:21621 JsSIP:RTCSession receiveInviteResponse() +8ms
jssip.js:21621 JsSIP:Dialog new UAC dialog created with status EARLY +2ms
jssip.js:21621 JsSIP:RTCSession session progress +2ms
sessions.jsx:139 Tryit: Session::setCallStatus()
sessions.jsx:284 Tryit: ChatBox::componentDidUpdate()
sessions.jsx:46 Tryit: Session::componentDidUpdate()
jssip.js:21621 JsSIP:Transport received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WS sb9dhbvefrf4.invalid;branch=z9hG4bK6330132;received=192.168.1.207;rport=31060
From: "2002" <sip:2002@192.168.1.111>;tag=omj9k66u8n
To: <sip:2001@192.168.1.111>;tag=as3868ee20
Call-ID: a5qndau4t06rbms2k65l
CSeq: 870 INVITE
Server: Asterisk PBX 13.4.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:2001@192.168.1.111:5060;transport=WS>
Content-Type: application/sdp
Require: timer
Content-Length: 2744
v=0
o=root 1799075026 1799075026 IN IP4 192.168.1.111
s=Asterisk PBX 13.4.0
c=IN IP4 192.168.1.111
b=CT:384
t=0 0
m=audio 14268 RTP/SAVPF 0 8 9 111 4 3 111 112 5 10 118 7 18 110 117 119 97 102 115 116 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
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:119 speex/32000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
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:126 telephone-event/8000
a=fmtp:126 0-16
a=maxptime:20
a=ice-ufrag:02b1f25b785eaa4f5b3bdef57a43b3b7
a=ice-pwd:0484653d5a7c6ae4626080072975b3d8
a=candidate:Hc0a8016f 1 UDP 2130706431 192.168.1.111 14268 typ host
a=candidate:Hc0a87a01 1 UDP 2130706431 192.168.122.1 14268 typ host
a=candidate:Sca418a86 1 UDP 1694498815 202.65.138.134 14268 typ srflx raddr 192.168.1.111 rport 14268
a=candidate:Hc0a8016f 2 UDP 2130706430 192.168.1.111 14269 typ host
a=candidate:Hc0a87a01 2 UDP 2130706430 192.168.122.1 14269 typ host
a=candidate:Sca418a86 2 UDP 1694498814 202.65.138.134 14269 typ srflx raddr 192.168.1.111 rport 14269
a=connection:new
a=setup:active
a=fingerprint:SHA-256 F9:BA:AC:66:E8:B0:8F:A0:3B:3A:8F:66:6D:3C:2B:E1:0A:F7:74:E5:C2:01:01:41:59:21:6B:27:72:F5:21:32
a=sendrecv
m=video 17794 RTP/SAVPF 100 31 34 98 99 104
a=ice-ufrag:6c0012fd5cea21b9799fa94711b89a1f
a=ice-pwd:3b3da9c74f654ab47da1373e2dacf792
a=candidate:Hc0a8016f 1 UDP 2130706431 192.168.1.111 17794 typ host
a=candidate:Hc0a87a01 1 UDP 2130706431 192.168.122.1 17794 typ host
a=candidate:Sca418a86 1 UDP 1694498815 202.65.138.134 17794 typ srflx raddr 192.168.1.111 rport 17794
a=candidate:Hc0a8016f 2 UDP 2130706430 192.168.1.111 17795 typ host
a=candidate:Hc0a87a01 2 UDP 2130706430 192.168.122.1 17795 typ host
a=candidate:Sca418a86 2 UDP 1694498814 202.65.138.134 17795 typ srflx raddr 192.168.1.111 rport 17795
a=connection:new
a=setup:active
a=fingerprint:SHA-256 F9:BA:AC:66:E8:B0:8F:A0:3B:3A:8F:66:6D:3C:2B:E1:0A:F7:74:E5:C2:01:01:41:59:21:6B:27:72:F5:21:32
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=sendrecv
+12s
jssip.js:21621 JsSIP:RTCSession receiveInviteResponse() +10ms
jssip.js:21621 JsSIP:Dialog dialog a5qndau4t06rbms2k65lomj9k66u8nas3868ee20 changed to CONFIRMED state +0ms
jssip.js:21621 rtcninja:RTCPeerConnection setRemoteDescription() +0ms
jssip.js:21621 rtcninja:RTCPeerConnection onsignalingstatechange() | signalingState: stable +15ms
jssip.js:21621 rtcninja:RTCPeerConnection setRemoteDescription() | success +5ms
jssip.js:21621 JsSIP:RTCSession session accepted +1ms
sessions.jsx:139 Tryit: Session::setCallStatus()
sessions.jsx:284 Tryit: ChatBox::componentDidUpdate()
sessions.jsx:46 Tryit: Session::componentDidUpdate()
jssip.js:21621 rtcninja:RTCPeerConnection getLocalStreams() +4ms
jssip.js:21621 rtcninja:RTCPeerConnection getLocalStreams() +0ms
jssip.js:21621 JsSIP:RTCSession sendRequest() +1ms
jssip.js:21621 JsSIP:RTCSession:Request new | ACK +0ms
jssip.js:21621 JsSIP:Transport sending WebSocket message:
ACK sip:2001@192.168.1.111:5060;transport=ws SIP/2.0
Via: SIP/2.0/WS sb9dhbvefrf4.invalid;branch=z9hG4bK4447000
Max-Forwards: 69
To: <sip:2001@192.168.1.111>;tag=as3868ee20
From: "2002" <sip:2002@192.168.1.111>;tag=omj9k66u8n
Call-ID: a5qndau4t06rbms2k65l
CSeq: 870 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: outbound
User-Agent: JsSIP 0.6.34
Content-Length: 0
+3ms
jssip.js:21621 JsSIP:RTCSession session confirmed +0ms
jssip.js:21621 rtcninja:RTCPeerConnection onaddstream() | stream: MediaStream +1ms
gui.js:240 Tryit: addstream()
jssip.js:21621 rtcninja:RTCPeerConnection oniceconnectionstatechange() | iceConnectionState: checking +2ms
jssip.js:21621 JsSIP:Transport received WebSocket text message:
BYE sip:sav0a3ef@sb9dhbvefrf4.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 192.168.1.111:5060;branch=z9hG4bK23871a50;rport
Max-Forwards: 70
From: <sip:2001@192.168.1.111>;tag=as3868ee20
To: "2002" <sip:2002@192.168.1.111>;tag=omj9k66u8n
Call-ID: a5qndau4t06rbms2k65l
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.4.0
Proxy-Authorization: Digest username="2002", realm="192.168.1.111", algorithm=MD5, uri="sip:192.168.1.111", nonce="014cb58e", response="08c27d6540be0f5665770f4825909e25"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
+3ms
jssip.js:21621 JsSIP:RTCSession receiveRequest() +7ms
jssip.js:21621 JsSIP:Transport sending WebSocket message:
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.1.111:5060;branch=z9hG4bK23871a50;rport
To: "2002" <sip:2002@192.168.1.111>;tag=omj9k66u8n
From: <sip:2001@192.168.1.111>;tag=as3868ee20
Call-ID: a5qndau4t06rbms2k65l
CSeq: 102 BYE
Supported: outbound
Content-Length: 0
+1ms
jssip.js:21621 JsSIP:RTCSession session ended +0ms
jssip.js:21621 JsSIP:RTCSession close() +0ms
jssip.js:21621 rtcninja:RTCPeerConnection close() +1ms
jssip.js:21621 JsSIP:RTCSession close() | closing local MediaStream +2ms
jssip.js:21621 rtcninja:Adapter closeMediaStream() | calling stop() on all the MediaStreamTrack +0ms
jssip.js:21621 JsSIP:Dialog dialog a5qndau4t06rbms2k65lomj9k66u8nas3868ee20 deleted +1ms
sessions.jsx:139 Tryit: Session::setCallStatus()
sessions.jsx:284 Tryit: ChatBox::componentDidUpdate()
sessions.jsx:46 Tryit: Session::componentDidUpdate()
jssip.js:21621 rtcninja:Adapter closeMediaStream() | calling stop() on all the MediaStreamTrack +4ms
gui.js:52 Tryit: removeSession
iscomposing.js:1146 iscomposing:CompositionIndicator close() +14s
iscomposing.js:1146 iscomposing:Composer setStatus() | from IDLE to IDLE +1ms
iscomposing.js:1146 iscomposing:Receiver setStatus() | from IDLE to IDLE +0ms
gui.js:89 Tryit: renderSessions
jssip.js:21621 JsSIP:NonInviteServerTransaction Timer J expired for transaction z9hG4bK23871a50 +4ms
jssip.js:21621 rtcninja:RTCPeerConnection oniceconnectionstatechange() | iceConnectionState: closed +0ms
jssip.js:21621 rtcninja:RTCPeerConnection onsignalingstatechange() | signalingState: closed +1ms
jssip.js:21621 JsSIP:InviteClientTransaction Timer B expired for transaction z9hG4bK6330132 +19s
jssip.js:21621 JsSIP:Transport received WebSocket text message:
I am also including some configuration files in case someone would find them helpful
- sip.conf
[general]
context=WebRTCContext
allowoverlap=no
realm=192.168.1.111
udpbindaddr=0.0.0.0:5060
transport=ws,udp
srvlookup=no
dtmfmode = rfc2833
videosupport=yes
textsupport=yes
nat = yes
media_address = 192.168.1.111
icesupport = yes
directmedia=no
encryption=yes
avpf=yes
force_avp=yes
rtcachefriends=no
dtlsenable = yes
dtlsverify = no
dtlscertfile = /etc/asterisk/keys/asterisk.pem
dtlsprivatekey = /etc/asterisk/keys/asterisk.pem
dtlscipher = tlsv1 ; <SSL cipher string>
dtlscafile = /etc/asterisk/keys/ca.crt
dtlssetup = actpass
[2001]
type=peer
username=2001 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=2001
callerid = "2001"<2001> ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=WebRTCContext ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=ws,udp ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass
disallow=all ; reset which voice codecs this device will accept or offer
allow=all ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
[2002]
type=peer
username=2002 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=2002
callerid = "2002"<2002> ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=WebRTCContext ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=ws,udp ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass
disallow=all ; reset which voice codecs this device will accept or offer
allow=all ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
2. extensions.conf (The extension I am using are at the bottom called from “WebRTCContext”)
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command "dialplan save" too
;
writeprotect=no
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass@provider
;FREENUMDOMAIN=mydomain.com ; domain to send on outbound
; freenum calls (uses outbound-freenum
; context)
[iaxprovider]
[trunklocal]
[trunktollfree]
[international]
[longdistance]
;include => trunkld
[local]
[outbound-freenum]
; We'll add more digits as needed. The purpose is to dial things
; like extension numbers at domains (ITAD number) so we're matching
; on lengths of 1 through 6 prior to the separator (the asterisk [*])
;
;exten => _X*X!,1,Goto(outbound-freenum2,${EXTEN},1)
;exten => _XX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
;exten => _XXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
;exten => _XXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
;exten => _XXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
;exten => _XXXXXX*X!,1,Goto(outbound-freenum2,${EXTEN},1)
[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten => _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same => n,Set(SUFFIX=${CUT(EXTEN,*,2-)}) ; make sure the suffix is all digits as well
same => n,GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1)
; filter out bad characters per the README-SERIOUSLY.best-practices.txt document
same => n,Set(TIMEOUT(absolute)=10800)
same => n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ; perform our lookup with freenum.org
same => n,GotoIf($["${isnresult}" != ""]?from)
same => n,Set(DIALSTATUS=CONGESTION)
same => n,Goto(fn-CONGESTION,1)
same => n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same => n,GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) ; check if we set the FREENUMDOMAIN global variable in [global]
same => n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ; if we did set it, then we'll use it for our outbound dialing domain
same => n(dial),Dial(SIP/${isnresult},40)
same => n,Goto(fn-${DIALSTATUS},1)
exten => fn-BUSY,1,Busy()
exten => _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same => n,Congestion()
[macro-trunkdial]
;exten => _s-.,1,NoOp
[stdexten]
;
[demo]
;include => stdexten
;
[page]
exten => _X.,1,Macro(page,SIP/${EXTEN})
;[mainmenu]
[public]
;
; ATTENTION: If your Asterisk is connected to the internet and you do
; not have allowguest=no in sip.conf, everybody out there may use your
; public context without authentication. In that case you want to
; double check which services you offer to the world.
;
include => demo
[default]
;
; By default we include the demo. In a production system, you
; probably don't want to have the demo there.
;
include => demo
[time]
[ani]
;--------------------------------------------------------------------------------------------------------
[LocalSets]
exten => 1001,1,Dial(SIP/Zoiper1,20)
exten => 1001,2,VoiceMail(501@default)
exten => 1001,n,HangUp()
exten => 1002,1,Dial(SIP/Zoiper2,20)
exten => 1002,2,VoiceMail(501@default)
exten => 1002,n,HangUp()
; --------------------------------------------
; -----WebRTC peers------------------------
[WebRTCContext]
exten => 2001,1,Dial(SIP/2001,20) ;
exten => 2002,1,Dial(SIP/2002,20) ;
exten => 1443,1,Dial(SIP/1443) ;
exten => 1444,1,Dial(SIP/1444) ;- rtp.conf
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
;
; Whether to enable or disable UDP checksums on RTP traffic
;
rtpchecksums=no
;
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)
;
;dtmftimeout=3000
; rtcpinterval = 5000 ; Milliseconds between rtcp reports
;(min 500, max 60000, default 5000)
;
; Enable strict RTP protection. This will drop RTP packets that
; do not come from the source of the RTP stream. This option is
; enabled by default.
strictrtp=no
;
; Number of packets containing consecutive sequence values needed
; to change the RTP source socket address. This option only comes
; into play while using strictrtp=yes. Consider changing this value
; if rtp packets are dropped from one or both ends after a call is
; connected. This option is set to 4 by default.
; probation=8
;
; Whether to enable or disable ICE support. This option is disabled by default.
icesupport=true
;
; Hostname or address for the STUN server used when determining the external
; IP address and port an RTP session can be reached at. The port number is
; optional. If omitted the default value of 3478 will be used. This option is
; disabled by default.
;
; e.g. stundaddr=mystun.server.com:3478
;
stunaddr=stun.l.google.com:19302
;
; Hostname or address for the TURN server to be used as a relay. The port
; number is optional. If omitted the default value of 3478 will be used.
; This option is disabled by default.
;
; e.g. turnaddr=myturn.server.com:34780
;
; turnaddr=
;
; Username used to authenticate with TURN relay server.
; turnusername=
;
; Password used to authenticate with TURN relay server.
; turnpassword=