# WebRTC Sound Problems **URL:** \ **Category:** Asterisk Support\ **Created:** [April 24, 2014, 3:34am UTC](https://community.asterisk.org/t/webrtc-sound-problems/45983 "2014-04-24T03:34:08Z")\ **Posts on this page:** 7\ **Page:** 1
**Author:** ![gfesj](https://avatars.discourse-cdn.com/v4/letter/g/a4c791/32.png) [@gfesj](https://community.asterisk.org/u/gfesj)\ **Post date:** [April 24, 2014, 3:34am UTC](https://community.asterisk.org/t/webrtc-sound-problems/45983/1 "2014-04-24T03:34:08Z")
Hi, my name is Gerald and I am try to use JSSIP and WebRTC, but we do not receive audio from calls. My instalation is CentOS 6.5 (Linux mercurio 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux) with Asterisk 11.9.0 built by root @ mercurio on a i686 running Linux on 2014-04-23 22:24:19 UTC. I configure all sip.conf/rtp.conf/htp.conf, I can register but when I try to make a call it show me at the SIP debug \<— SIP read from WS:189.232.221.54:63589 —\> BYE sip:1000@187.122.82.197:0;transport=ws SIP/2.0 Via: SIP/2.0/WS mmj0ua0tjbct.invalid;branch=z9hG4bK1767708 Max-Forwards: 69 To: sip:1000@189.232.221.54;tag=as1514d3d2 From: “G” sip:8001@189.232.221.54;tag=c0ujquk1fj Call-ID: 6n90l9o94cd2b5bcltbk CSeq: 7553 BYE Reason: SIP ;cause=488; text="Not Acceptable Here" Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 I would like some help to solve this issue! ---
**Author:** ![navaismo](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/navaismo/32/2481_2.png) [@navaismo](https://community.asterisk.org/u/navaismo)\ **Post date:** [April 24, 2014, 4:47am UTC](https://community.asterisk.org/t/webrtc-sound-problems/45983/2 "2014-04-24T04:47:28Z")
Not acceptable here using webrtc usually is caused by a misconfiguration in the sip peer. You need to provide the real configuration for SIP, JSSIP and the debugs from asterisk(sip debug) and web browser(JavaScript debug). ---
**Author:** ![gfesj](https://avatars.discourse-cdn.com/v4/letter/g/a4c791/32.png) [@gfesj](https://community.asterisk.org/u/gfesj)\ **Post date:** [April 24, 2014, 10:03pm UTC](https://community.asterisk.org/t/webrtc-sound-problems/45983/3 "2014-04-24T22:03:15Z")
Thank you for the help, here are the files. SIP.CONF [code]; SIP Configuration example for Asterisk [general] context=public ; Default context for incoming calls. Defaults to 'default’ allowoverlap=no ; Disable overlap dialing support. (Default is yes) [realm=webrtc.no-ip.org](http://realm=webrtc.no-ip.org) ; Realm for digest authentication udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) tcpenable=no ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) transport=udp,ws ; Set the default transports. The order determines the primary default transport. ; If tcpenable=no and the transport set is tcp, we will fallback to UDP. srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ; Specifying a port in a SIP peer definition or ; when dialing outbound calls will supress SRV ; lookups for that peer or call. localnet=192.168.192.7/255.255.255.0 [externaddr=webrtc.no-ip.org](http://externaddr=webrtc.no-ip.org) nat=auto\_force\_rport,auto\_comedia directmedia=yes ; Asterisk by default tries to redirect the [1234] type=friend regexten=1234 ; When they register, create extension 1234 secret=\_\_\_\_\_\_ callerid=“AAAAAA” \<1234\> nat=force\_rport,comedia host=dynamic ; This device needs to register directmedia=no ; Typically set to NO if behind NAT disallow=all allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw allow=alaw registertrying=yes ; Send a 10 [8001] context=default trustrpid=yes sendrpid=no qualify=yes qualifyfreq=600 type=friend ; we only want to call out, not be called regexten=8001 secret=\_\_\_\_\_ nat=auto\_force\_rport,auto\_comedia encryption=yes remotesecret=\_\_\_\_\_ defaultuser=8001 ; Authentication user for outbound proxies fromuser=8001 ; Many SIP providers require this! host=dynamic avpf=yes icesupport=yes directmedia=no dial=SIP/8001 disallow=all allow=ulaw [/code] SIP.LOG [code]\<— SIP read from WS:189.232.221.54:56882 —\> REGISTER sip:webrtc.no-ip.org SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8958760 Max-Forwards: 69 To: sip:8001@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7 Call-ID: 1jtu7lhr09rhmilv2k6sgu CSeq: 81 REGISTER Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:a499f638-557f-45c2-87ab-52feb7df49b1”;expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 \<-------------\> \<— Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8958760;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7 To: sip:8001@webrtc.no-ip.org;tag=as05e0dab9 Call-ID: 1jtu7lhr09rhmilv2k6sgu CSeq: 81 REGISTER Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce="5a4ecdb4" Content-Length: 0 \<------------\> \<— SIP read from WS:189.232.221.54:56882 —\> REGISTER sip:webrtc.no-ip.org SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8231095 Max-Forwards: 69 To: sip:8001@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7 Call-ID: 1jtu7lhr09rhmilv2k6sgu CSeq: 82 REGISTER Authorization: Digest algorithm=MD5, username=“8001”, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce=“5a4ecdb4”, uri=“sip:webrtc.no-ip.org”, response="8df3a2d05686d9bc7bc2b856bbe5fba0" Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:a499f638-557f-45c2-87ab-52feb7df49b1”;expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 \<-------------\> OPTIONS sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK07255c34 Max-Forwards: 70 From: “asterisk” sip:8001@192.168.192.7;tag=as42109a96 To: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws Contact: sip:8001@192.168.192.7:5060;transport=WS Call-ID: 176f4dc53b07ee792a01aa9e415bb16a@192.168.192.7:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.7.0 Date: Thu, 24 Apr 2014 20:58:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 * * * \<— Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8231095;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7 To: sip:8001@webrtc.no-ip.org;tag=as05e0dab9 Call-ID: 1jtu7lhr09rhmilv2k6sgu CSeq: 82 REGISTER Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 600 Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;expires=600 Date: Thu, 24 Apr 2014 20:58:57 GMT Content-Length: 0 \<------------\> [Khaxixe\*CLI\> [0KScheduling destruction of SIP dialog ‘1jtu7lhr09rhmilv2k6sgu’ in 32000 ms (Method: REGISTER) \<— SIP read from WS:189.232.221.54:56882 —\> SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK07255c34 To: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;tag=p9g6uh7jar From: “asterisk” sip:8001@192.168.192.7;tag=as42109a96 Call-ID: 176f4dc53b07ee792a01aa9e415bb16a@192.168.192.7:5060 CSeq: 102 OPTIONS Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Accept: application/sdp,application/dtmf-relay Content-Length: 0 \<-------------\> \<— SIP read from WS:189.232.221.54:56882 —\> INVITE [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6705 INVITE Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;ob Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 2238 v=0 o=- 7089255009147564118 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm m=audio 57129 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.101.103 a=rtcp:57129 IN IP4 192.168.101.103 a=candidate:1117122343 1 udp 2122260223 192.168.101.103 57129 typ host generation 0 a=candidate:1117122343 2 udp 2122260223 192.168.101.103 57129 typ host generation 0 a=candidate:1755041049 1 udp 2122194687 192.168.74.1 57130 typ host generation 0 a=candidate:1755041049 2 udp 2122194687 192.168.74.1 57130 typ host generation 0 a=candidate:4254299990 1 udp 2122129151 10.246.40.1 57131 typ host generation 0 a=candidate:4254299990 2 udp 2122129151 10.246.40.1 57131 typ host generation 0 a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=ice-ufrag:0s4bNH9Jo66GKXst a=ice-pwd:3cBHEENG5P3FvFiQ5DFUxiwC a=ice-options:google-ice a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:YpwqkvWMSZ2YMGcNajJ4j70FhJQDEZ1d3n3cMUMf a=crypto:1 AES\_CM\_128\_HMAC\_SHA1\_80 inline:8IE6Vq/KBMXpJUCAqUxN9CD+A2wM751sik9nWU7M a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 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:112670393 cname:rahmGKoZTz8GDKNr a=ssrc:112670393 msid:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm 5a7e9bc3-5165-4303-9c51-6ce2feeef02a a=ssrc:112670393 mslabel:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm a=ssrc:112670393 label:5a7e9bc3-5165-4303-9c51-6ce2feeef02a \<-------------\> — (13 headers 47 lines) — Using INVITE request as basis request - 73c6ks5m99l72uddthv6 Found peer ‘8001’ for ‘8001’ from 189.232.221.54:56882 \<— Reliably Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 To: sip:1000@webrtc.no-ip.org;tag=as52137f59 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6705 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce="04f22c86" Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog ‘73c6ks5m99l72uddthv6’ in 17472 ms (Method: INVITE) \<— SIP read from WS:189.232.221.54:56882 —\> ACK [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450 To: sip:1000@webrtc.no-ip.org;tag=as52137f59 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6705 ACK \<-------------\> — (6 headers 0 lines) — \<— SIP read from WS:189.232.221.54:56882 —\> INVITE [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8137428 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6706 INVITE Authorization: Digest algorithm=MD5, username=“8001”, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce=“04f22c86”, [uri="sip:1000@webrtc.no-ip.org](mailto:uri=%22sip:1000@webrtc.no-ip.org)", response="651e8183d2aa4335c625fbe88131efe7" Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;ob Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 2238 v=0 o=- 7089255009147564118 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm m=audio 57129 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.101.103 a=rtcp:57129 IN IP4 192.168.101.103 a=candidate:1117122343 1 udp 2122260223 192.168.101.103 57129 typ host generation 0 a=candidate:1117122343 2 udp 2122260223 192.168.101.103 57129 typ host generation 0 a=candidate:1755041049 1 udp 2122194687 192.168.74.1 57130 typ host generation 0 a=candidate:1755041049 2 udp 2122194687 192.168.74.1 57130 typ host generation 0 a=candidate:4254299990 1 udp 2122129151 10.246.40.1 57131 typ host generation 0 a=candidate:4254299990 2 udp 2122129151 10.246.40.1 57131 typ host generation 0 a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=ice-ufrag:0s4bNH9Jo66GKXst a=ice-pwd:3cBHEENG5P3FvFiQ5DFUxiwC a=ice-options:google-ice a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:YpwqkvWMSZ2YMGcNajJ4j70FhJQDEZ1d3n3cMUMf a=crypto:1 AES\_CM\_128\_HMAC\_SHA1\_80 inline:8IE6Vq/KBMXpJUCAqUxN9CD+A2wM751sik9nWU7M a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 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:112670393 cname:rahmGKoZTz8GDKNr a=ssrc:112670393 msid:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm 5a7e9bc3-5165-4303-9c51-6ce2feeef02a a=ssrc:112670393 mslabel:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm a=ssrc:112670393 label:5a7e9bc3-5165-4303-9c51-6ce2feeef02a \<-------------\> — (14 headers 47 lines) — Using INVITE request as basis request - 73c6ks5m99l72uddthv6 Found peer ‘8001’ for ‘8001’ from 189.232.221.54:56882 Found RTP audio format 111 Found RTP audio format 103 Found RTP audio format 104 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 unknown media 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 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 Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.101.103:57129 Looking for 1000 in default (domain [webrtc.no-ip.org](http://webrtc.no-ip.org)) list\_route: hop: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;ob \<— Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8137428;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 To: sip:1000@webrtc.no-ip.org Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6706 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:1000@192.168.192.7:5060;transport=WS Content-Length: 0 \<------------\> OKAudio is at 14566 0KAdding codec 100003 (ulaw) to SDP 0KAdding non-codec 0x1 (telephone-event) to SDP 0K \<— Reliably Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8137428;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 To: sip:1000@webrtc.no-ip.org;tag=as37deb063 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6706 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:1000@192.168.192.7:5060;transport=WS Content-Type: application/sdp Content-Length: 321 v=0 o=root 385143318 385143318 IN IP4 192.168.192.7 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.192.7 t=0 0 m=audio 14566 RTP/SAVPF 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:uSNE9caBZA2PPo6wPPWYkco5RrYdwU+jZyWdMXfE \<------------\> \<— SIP read from WS:189.232.221.54:56882 —\> ACK sip:1000@192.168.192.7:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5964741 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org;tag=as37deb063 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6706 ACK Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 \<-------------\> — (10 headers 0 lines) — \<— SIP read from WS:189.232.221.54:56882 —\> BYE sip:1000@192.168.192.7:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK4512882 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org;tag=as37deb063 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6707 BYE Reason: SIP ;cause=488; text="Not Acceptable Here" Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 \<-------------\> — (11 headers 0 lines) — Scheduling destruction of SIP dialog ‘73c6ks5m99l72uddthv6’ in 17472 ms (Method: BYE) \<— Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK4512882;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 To: sip:1000@webrtc.no-ip.org;tag=as37deb063 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6707 BYE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 \<------------\> [/code] JavaSCRIPTDebug.LOG [code]GET [http://tryit.jssip.net/js/custom.js](http://tryit.jssip.net/js/custom.js) 404 (Not Found) [tryit.jssip.net/:22](http://tryit.jssip.net/:22) event.returnValue is deprecated. Please use the standard event.preventDefault() instead. jquery-1.6.1.min.js:17 JsSIP | UA | configuration parameters after validation: jssip-devel.js:5783 • via\_host: “1lhjsfc8k672.invalid” jssip-devel.js:5794 • password: NOT SHOWN jssip-devel.js:5791 • register\_expires: 600 jssip-devel.js:5794 • register\_min\_expires: 120 jssip-devel.js:5794 • register: true jssip-devel.js:5794 • registrar\_server: sip:webrtc.no-ip.org jssip-devel.js:5788 • ws\_server\_max\_reconnection: 3 jssip-devel.js:5794 • ws\_server\_reconnection\_timeout: 4 jssip-devel.js:5794 • connection\_recovery\_min\_interval: 2 jssip-devel.js:5794 • connection\_recovery\_max\_interval: 30 jssip-devel.js:5794 • use\_preloaded\_route: false jssip-devel.js:5794 • no\_answer\_timeout: 60000 jssip-devel.js:5794 • stun\_servers: [“stun:74.125.132.127:19302”] jssip-devel.js:5794 • turn\_servers: [] jssip-devel.js:5794 • trace\_sip: true jssip-devel.js:5794 • hack\_via\_tcp: false jssip-devel.js:5794 • hack\_ip\_in\_contact: false jssip-devel.js:5794 • uri: [sip:8001@webrtc.no-ip.org](mailto:sip:8001@webrtc.no-ip.org) jssip-devel.js:5788 • ws\_servers: [{“ws\_uri”:“ws://webrtc.no-ip.org:8088/ws”,“sip\_uri”:“sip:webrtc.no-ip.org:8088;transport=ws;lr”,“weight”:0,“status”:0,“scheme”:“WS”}] jssip-devel.js:5794 • display\_name: “G” jssip-devel.js:5794 • instance\_id: “a499f638-557f-45c2-87ab-52feb7df49b1” jssip-devel.js:5794 • jssip\_id: “73c6k” jssip-devel.js:5794 • hostport\_params: “[webrtc.no-ip.org](http://webrtc.no-ip.org)” jssip-devel.js:5794 • authorization\_user: “8001” jssip-devel.js:5794 JsSIP | EVENT EMITTER | adding event newMessage jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event newRTCSession jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event registrationFailed jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event unregistered jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event registered jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event disconnected jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event connected jssip-devel.js:67 JsSIP | EVENT EMITTER | new listener added to event connected jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event disconnected jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event newRTCSession jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event newMessage jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event registered jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event unregistered jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event registrationFailed jssip-devel.js:63 JsSIP | UA | user requested startup… jssip-devel.js:5238 JsSIP | TRANSPORT | connecting to WebSocket ws://webrtc.no-ip.org:8088/ws jssip-devel.js:568 JsSIP | TRANSPORT | WebSocket ws://webrtc.no-ip.org:8088/ws connected jssip-devel.js:604 JsSIP | UA | connection state set to 0 jssip-devel.js:5360 JsSIP | EVENT EMITTER | emitting event connected jssip-devel.js:187 JsSIP | TRANSPORT | sending WebSocket message: REGISTER sip:webrtc.no-ip.org SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8958760 Max-Forwards: 69 To: sip:8001@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7 Call-ID: 1jtu7lhr09rhmilv2k6sgu CSeq: 81 REGISTER Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:a499f638-557f-45c2-87ab-52feb7df49b1”;expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 jssip-devel.js:519 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8958760;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7 To: sip:8001@webrtc.no-ip.org;tag=as05e0dab9 Call-ID: 1jtu7lhr09rhmilv2k6sgu CSeq: 81 REGISTER Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce="5a4ecdb4" Content-Length: 0 jssip-devel.js:686 JsSIP | TRANSPORT | sending WebSocket message: REGISTER sip:webrtc.no-ip.org SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8231095 Max-Forwards: 69 To: sip:8001@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7 Call-ID: 1jtu7lhr09rhmilv2k6sgu CSeq: 82 REGISTER Authorization: Digest algorithm=MD5, username=“8001”, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce=“5a4ecdb4”, uri=“sip:webrtc.no-ip.org”, response="8df3a2d05686d9bc7bc2b856bbe5fba0" Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:a499f638-557f-45c2-87ab-52feb7df49b1”;expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 jssip-devel.js:519 JsSIP | TRANSPORT | received WebSocket text message: OPTIONS sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK07255c34 Max-Forwards: 70 From: “asterisk” sip:8001@192.168.192.7;tag=as42109a96 To: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws Contact: sip:8001@192.168.192.7:5060;transport=WS Call-ID: 176f4dc53b07ee792a01aa9e415bb16a@192.168.192.7:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.7.0 Date: Thu, 24 Apr 2014 20:58:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 jssip-devel.js:686 JsSIP | TRANSPORT | sending WebSocket message: SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK07255c34 To: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;tag=p9g6uh7jar From: “asterisk” sip:8001@192.168.192.7;tag=as42109a96 Call-ID: 176f4dc53b07ee792a01aa9e415bb16a@192.168.192.7:5060 CSeq: 102 OPTIONS Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Accept: application/sdp,application/dtmf-relay Content-Length: 0 jssip-devel.js:519 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8231095;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=ivaakt2cc7 To: sip:8001@webrtc.no-ip.org;tag=as05e0dab9 Call-ID: 1jtu7lhr09rhmilv2k6sgu CSeq: 82 REGISTER Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 600 Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;expires=600 Date: Thu, 24 Apr 2014 20:58:57 GMT Content-Length: 0 jssip-devel.js:686 JsSIP | EVENT EMITTER | emitting event registered jssip-devel.js:187 JsSIP | TRANSACTION | Timer J expired for non-INVITE server transaction z9hG4bK07255c34 jssip-devel.js:2094 Registered init.js:428 JsSIP | EVENT EMITTER | adding event newDTMF jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event ended jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event started jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event failed jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event progress jssip-devel.js:67 JsSIP | EVENT EMITTER | emitting event newRTCSession jssip-devel.js:187 JsSIP | RTC SESSION | requesting access to local media jssip-devel.js:3442 JsSIP | EVENT EMITTER | new listener added to event progress jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event started jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event failed jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event newDTMF jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event ended jssip-devel.js:63 JsSIP | RTC SESSION | got local media stream jssip-devel.js:3446 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1117122343 1 udp 2122260223 192.168.101.103 57129 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1117122343 2 udp 2122260223 192.168.101.103 57129 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1755041049 1 udp 2122194687 192.168.74.1 57130 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1755041049 2 udp 2122194687 192.168.74.1 57130 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:4254299990 1 udp 2122129151 10.246.40.1 57131 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:4254299990 2 udp 2122129151 10.246.40.1 57131 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0 jssip-devel.js:3401 JsSIP | TRANSPORT | sending WebSocket message: INVITE [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6705 INVITE Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;ob Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 2238 v=0 o=- 7089255009147564118 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm m=audio 57129 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.101.103 a=rtcp:57129 IN IP4 192.168.101.103 a=candidate:1117122343 1 udp 2122260223 192.168.101.103 57129 typ host generation 0 a=candidate:1117122343 2 udp 2122260223 192.168.101.103 57129 typ host generation 0 a=candidate:1755041049 1 udp 2122194687 192.168.74.1 57130 typ host generation 0 a=candidate:1755041049 2 udp 2122194687 192.168.74.1 57130 typ host generation 0 a=candidate:4254299990 1 udp 2122129151 10.246.40.1 57131 typ host generation 0 a=candidate:4254299990 2 udp 2122129151 10.246.40.1 57131 typ host generation 0 a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=ice-ufrag:0s4bNH9Jo66GKXst a=ice-pwd:3cBHEENG5P3FvFiQ5DFUxiwC a=ice-options:google-ice a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:YpwqkvWMSZ2YMGcNajJ4j70FhJQDEZ1d3n3cMUMf a=crypto:1 AES\_CM\_128\_HMAC\_SHA1\_80 inline:8IE6Vq/KBMXpJUCAqUxN9CD+A2wM751sik9nWU7M a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 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:112670393 cname:rahmGKoZTz8GDKNr a=ssrc:112670393 msid:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm 5a7e9bc3-5165-4303-9c51-6ce2feeef02a a=ssrc:112670393 mslabel:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm a=ssrc:112670393 label:5a7e9bc3-5165-4303-9c51-6ce2feeef02a jssip-devel.js:519 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 To: sip:1000@webrtc.no-ip.org;tag=as52137f59 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6705 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce="04f22c86" Content-Length: 0 jssip-devel.js:686 JsSIP | TRANSPORT | sending WebSocket message: ACK [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5326450 To: sip:1000@webrtc.no-ip.org;tag=as52137f59 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6705 ACK jssip-devel.js:519 JsSIP | TRANSPORT | sending WebSocket message: INVITE [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8137428 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6706 INVITE Authorization: Digest algorithm=MD5, username=“8001”, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce=“04f22c86”, [uri="sip:1000@webrtc.no-ip.org](mailto:uri=%22sip:1000@webrtc.no-ip.org)", response="651e8183d2aa4335c625fbe88131efe7" Contact: sip:u18vq2jc@1lhjsfc8k672.invalid;transport=ws;ob Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 2238 v=0 o=- 7089255009147564118 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm m=audio 57129 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.101.103 a=rtcp:57129 IN IP4 192.168.101.103 a=candidate:1117122343 1 udp 2122260223 192.168.101.103 57129 typ host generation 0 a=candidate:1117122343 2 udp 2122260223 192.168.101.103 57129 typ host generation 0 a=candidate:1755041049 1 udp 2122194687 192.168.74.1 57130 typ host generation 0 a=candidate:1755041049 2 udp 2122194687 192.168.74.1 57130 typ host generation 0 a=candidate:4254299990 1 udp 2122129151 10.246.40.1 57131 typ host generation 0 a=candidate:4254299990 2 udp 2122129151 10.246.40.1 57131 typ host generation 0 a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=ice-ufrag:0s4bNH9Jo66GKXst a=ice-pwd:3cBHEENG5P3FvFiQ5DFUxiwC a=ice-options:google-ice a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:YpwqkvWMSZ2YMGcNajJ4j70FhJQDEZ1d3n3cMUMf a=crypto:1 AES\_CM\_128\_HMAC\_SHA1\_80 inline:8IE6Vq/KBMXpJUCAqUxN9CD+A2wM751sik9nWU7M a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 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:112670393 cname:rahmGKoZTz8GDKNr a=ssrc:112670393 msid:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm 5a7e9bc3-5165-4303-9c51-6ce2feeef02a a=ssrc:112670393 mslabel:CqG5LhoCZlfsJzjS78Fo15VzrVZ8fn9HIwTm a=ssrc:112670393 label:5a7e9bc3-5165-4303-9c51-6ce2feeef02a jssip-devel.js:519 JsSIP | TRANSACTION | Timer D expired for INVITE client transaction z9hG4bK5326450 jssip-devel.js:1969 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 100 Trying Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8137428;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 To: sip:1000@webrtc.no-ip.org Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6706 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:1000@192.168.192.7:5060;transport=WS Content-Length: 0 jssip-devel.js:686 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK8137428;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 To: sip:1000@webrtc.no-ip.org;tag=as37deb063 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6706 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:1000@192.168.192.7:5060;transport=WS Content-Type: application/sdp Content-Length: 321 v=0 o=root 385143318 385143318 IN IP4 192.168.192.7 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.192.7 t=0 0 m=audio 14566 RTP/SAVPF 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:uSNE9caBZA2PPo6wPPWYkco5RrYdwU+jZyWdMXfE jssip-devel.js:686 JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-devel.js:2546 Failed to set remote answer sdp: Called with a SDP without ice-ufrag and ice-pwd. jssip-devel.js:4544 JsSIP | TRANSPORT | sending WebSocket message: ACK sip:1000@192.168.192.7:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK5964741 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org;tag=as37deb063 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6706 ACK Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 jssip-devel.js:519 JsSIP | TRANSPORT | sending WebSocket message: BYE sip:1000@192.168.192.7:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK4512882 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org;tag=as37deb063 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6707 BYE Reason: SIP ;cause=488; text="Not Acceptable Here" Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 jssip-devel.js:519 JsSIP | RTC SESSION | closing INVITE session 73c6ks5m99l72uddthv61s29gab6p9 jssip-devel.js:4248 JsSIP | RTC SESSION | closing PeerConnection jssip-devel.js:3424 JsSIP | DIALOG | dialog 73c6ks5m99l72uddthv61s29gab6p9as37deb063 deleted jssip-devel.js:2566 JsSIP | EVENT EMITTER | emitting event failed jssip-devel.js:187 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WS 1lhjsfc8k672.invalid;branch=z9hG4bK4512882;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=1s29gab6p9 To: sip:1000@webrtc.no-ip.org;tag=as37deb063 Call-ID: 73c6ks5m99l72uddthv6 CSeq: 6707 BYE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 [/code] ---
**Author:** ![navaismo](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/navaismo/32/2481_2.png) [@navaismo](https://community.asterisk.org/u/navaismo)\ **Post date:** [April 24, 2014, 11:18pm UTC](https://community.asterisk.org/t/webrtc-sound-problems/45983/4 "2014-04-24T23:18:00Z")
`Failed to set remote answer sdp: Called with a SDP without ice-ufrag and ice-pwd. jssip-devel.js:4544 JsSIP | TRANSPORT | sending WebSocket message:` Seems like your asterisk is not compiled with ICE Support install libuuid/uuid and libuuid-devel/uuid-devel and recompile asterisk then try again. ---
**Author:** ![gfesj](https://avatars.discourse-cdn.com/v4/letter/g/a4c791/32.png) [@gfesj](https://community.asterisk.org/u/gfesj)\ **Post date:** [April 25, 2014, 1:53am UTC](https://community.asterisk.org/t/webrtc-sound-problems/45983/5 "2014-04-25T01:53:54Z")
Thanks for the tip, it solved the connection problem, but still can not get audio on the web page. Here are the new log files. SIP.log [code]\<— SIP read from WS:189.232.221.54:59385 —\> REGISTER sip:XXXX.no-ip.org SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK4913074 Max-Forwards: 69 To: sip:8001@webrtc.no-ip.org From: “G” sip:8001@XXXX.no-ip.org;tag=lftr2g6uci Call-ID: m94e59t1sqbl2pobami4no CSeq: 81 REGISTER Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:1b399baa-f209-4868-9e1e-490fb50fb0ea”;expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 \<-------------\> \<— Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK4913074;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci To: sip:8001@webrtc.no-ip.org;tag=as320bbfd8 Call-ID: m94e59t1sqbl2pobami4no CSeq: 81 REGISTER Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce="15ee906d" Content-Length: 0 \<------------\> \<— SIP read from WS:189.232.221.54:59385 —\> REGISTER sip:webrtc.no-ip.org SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK8025886 Max-Forwards: 69 To: sip:8001@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci Call-ID: m94e59t1sqbl2pobami4no CSeq: 82 REGISTER Authorization: Digest algorithm=MD5, username=“8001”, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce=“15ee906d”, uri=“sip:webrtc.no-ip.org”, response="332f925df91f25dbfbaf82267a6c87d1" Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:1b399baa-f209-4868-9e1e-490fb50fb0ea”;expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 \<-------------\> Reliably Transmitting (no NAT) to 189.232.221.54:59385: OPTIONS sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK56a57557 Max-Forwards: 70 From: “asterisk” sip:8001@192.168.192.7;tag=as37fa2738 To: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws Contact: sip:8001@192.168.192.7:5060;transport=WS Call-ID: 63c9aa52286750c914d6d89250525a51@192.168.192.7:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.7.0 Date: Fri, 25 Apr 2014 01:44:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 * * * \<— Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK8025886;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci To: sip:8001@webrtc.no-ip.org;tag=as320bbfd8 Call-ID: m94e59t1sqbl2pobami4no CSeq: 82 REGISTER Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 600 Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;expires=600 Date: Fri, 25 Apr 2014 01:44:11 GMT Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog ‘m94e59t1sqbl2pobami4no’ in 32000 ms (Method: REGISTER) \<— SIP read from WS:189.232.221.54:59385 —\> SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK56a57557 To: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;tag=nljv5atrao From: “asterisk” sip:8001@192.168.192.7;tag=as37fa2738 Call-ID: 63c9aa52286750c914d6d89250525a51@192.168.192.7:5060 CSeq: 102 OPTIONS Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Accept: application/sdp,application/dtmf-relay Content-Length: 0 \<-------------\> \<— SIP read from WS:189.232.221.54:59385 —\> INVITE [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv Call-ID: cgu3s60etphsvr40mneg CSeq: 5919 INVITE Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;ob Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 2242 v=0 o=- 3652154159910496246 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS 5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 m=audio 54280 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.101.103 a=rtcp:54280 IN IP4 192.168.101.103 a=candidate:1117122343 1 udp 2122260223 192.168.101.103 54280 typ host generation 0 a=candidate:1117122343 2 udp 2122260223 192.168.101.103 54280 typ host generation 0 a=candidate:1755041049 1 udp 2122194687 192.168.74.1 54281 typ host generation 0 a=candidate:1755041049 2 udp 2122194687 192.168.74.1 54281 typ host generation 0 a=candidate:4254299990 1 udp 2122129151 10.246.40.1 54282 typ host generation 0 a=candidate:4254299990 2 udp 2122129151 10.246.40.1 54282 typ host generation 0 a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=ice-ufrag:eNWxE4gTTqnCVyMu a=ice-pwd:ZoVKNOv2HlfHE9VNyW3NSIAO a=ice-options:google-ice a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:tCMlVm6sT9MJMxbzKEmsDV1e3VVvNa5MwkUKPKzN a=crypto:1 AES\_CM\_128\_HMAC\_SHA1\_80 inline:qHzInMoGik8HNFMVgi6IjHdClreeyslHlw11nvn8 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 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:3951139034 cname:1GEOwdZBp2a0BbWt a=ssrc:3951139034 msid:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 9cda3eb5-5a5c-4cb3-ad6d-0a153309d435 a=ssrc:3951139034 mslabel:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 a=ssrc:3951139034 label:9cda3eb5-5a5c-4cb3-ad6d-0a153309d435 \<-------------\> — (13 headers 47 lines) — Using INVITE request as basis request - cgu3s60etphsvr40mneg Found peer ‘8001’ for ‘8001’ from 189.232.221.54:59385 \<— Reliably Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv To: sip:1000@webrtc.no-ip.org;tag=as0464893c Call-ID: cgu3s60etphsvr40mneg CSeq: 5919 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce="12671f92" Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog ‘cgu3s60etphsvr40mneg’ in 15552 ms (Method: INVITE) \<— SIP read from WS:189.232.221.54:59385 —\> ACK [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051 To: sip:1000@webrtc.no-ip.org;tag=as0464893c From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv Call-ID: cgu3s60etphsvr40mneg CSeq: 5919 ACK \<-------------\> — (6 headers 0 lines) — \<— SIP read from WS:189.232.221.54:59385 —\> INVITE [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv Call-ID: cgu3s60etphsvr40mneg CSeq: 5920 INVITE Authorization: Digest algorithm=MD5, username=“8001”, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce=“12671f92”, [uri="sip:1000@webrtc.no-ip.org](mailto:uri=%22sip:1000@webrtc.no-ip.org)", response="63ef359a7befb51ebef6a306a7262d82" Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;ob Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 2242 v=0 o=- 3652154159910496246 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS 5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 m=audio 54280 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.101.103 a=rtcp:54280 IN IP4 192.168.101.103 a=candidate:1117122343 1 udp 2122260223 192.168.101.103 54280 typ host generation 0 a=candidate:1117122343 2 udp 2122260223 192.168.101.103 54280 typ host generation 0 a=candidate:1755041049 1 udp 2122194687 192.168.74.1 54281 typ host generation 0 a=candidate:1755041049 2 udp 2122194687 192.168.74.1 54281 typ host generation 0 a=candidate:4254299990 1 udp 2122129151 10.246.40.1 54282 typ host generation 0 a=candidate:4254299990 2 udp 2122129151 10.246.40.1 54282 typ host generation 0 a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=ice-ufrag:eNWxE4gTTqnCVyMu a=ice-pwd:ZoVKNOv2HlfHE9VNyW3NSIAO a=ice-options:google-ice a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:tCMlVm6sT9MJMxbzKEmsDV1e3VVvNa5MwkUKPKzN a=crypto:1 AES\_CM\_128\_HMAC\_SHA1\_80 inline:qHzInMoGik8HNFMVgi6IjHdClreeyslHlw11nvn8 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 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:3951139034 cname:1GEOwdZBp2a0BbWt a=ssrc:3951139034 msid:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 9cda3eb5-5a5c-4cb3-ad6d-0a153309d435 a=ssrc:3951139034 mslabel:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 a=ssrc:3951139034 label:9cda3eb5-5a5c-4cb3-ad6d-0a153309d435 \<-------------\> — (14 headers 47 lines) — Using INVITE request as basis request - cgu3s60etphsvr40mneg Found peer ‘8001’ for ‘8001’ from 189.232.221.54:59385 Found RTP audio format 111 Found RTP audio format 103 Found RTP audio format 104 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 unknown media 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 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 Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.101.103:54280 Looking for 1000 in default (domain [webrtc.no-ip.org](http://webrtc.no-ip.org)) list\_route: hop: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;ob \<— Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv To: sip:1000@webrtc.no-ip.org Call-ID: cgu3s60etphsvr40mneg CSeq: 5920 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:1000@192.168.192.7:5060;transport=WS Content-Length: 0 \<------------\> Audio is at 16094 Adding codec 100003 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP \<— Reliably Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8 Call-ID: cgu3s60etphsvr40mneg CSeq: 5920 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:1000@192.168.192.7:5060;transport=WS Content-Type: application/sdp Content-Length: 827 v=0 o=root 1916711608 1916711608 IN IP4 192.168.192.7 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.192.7 t=0 0 m=audio 16094 RTP/SAVPF 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:508b3b3a55b6364115e87667717fe8b3 a=ice-pwd:12147891493792b1612c66d02e20cfd0 a=candidate:Hc0a8c007 1 UDP 2130706431 192.168.192.7 16094 typ host a=candidate:Hc0a80103 1 UDP 2130706431 192.168.1.3 16094 typ host a=candidate:Sc953b8ef 1 UDP 1694498815 201.83.184.239 10332 typ srflx a=candidate:Hc0a8c007 2 UDP 2130706430 192.168.192.7 16095 typ host a=candidate:Hc0a80103 2 UDP 2130706430 192.168.1.3 16095 typ host a=candidate:Sc953b8ef 2 UDP 1694498814 201.83.184.239 10332 typ srflx a=sendrecv a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:5HrqW3GMP29G3i/zeH2c6B1DTbOS0UNdeeuY6i/f \<------------\> \<— SIP read from WS:189.232.221.54:59385 —\> ACK sip:1000@192.168.192.7:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3660144 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv Call-ID: cgu3s60etphsvr40mneg CSeq: 5920 ACK Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 \<-------------\> — (10 headers 0 lines) — Really destroying SIP dialog ‘m94e59t1sqbl2pobami4no’ Method: REGISTER \<— SIP read from WS:189.232.221.54:59385 —\> BYE sip:1000@192.168.192.7:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3459967 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv Call-ID: cgu3s60etphsvr40mneg CSeq: 5921 BYE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 \<-------------\> — (10 headers 0 lines) — Scheduling destruction of SIP dialog ‘cgu3s60etphsvr40mneg’ in 15552 ms (Method: BYE) \<— Transmitting (no NAT) to 189.232.221.54:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3459967;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8 Call-ID: cgu3s60etphsvr40mneg CSeq: 5921 BYE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 \<------------\> Really destroying SIP dialog ‘cgu3s60etphsvr40mneg’ Method: BYE [/code] JavaScriptDebug.log [code]GET [http://tryit.jssip.net/js/custom.js](http://tryit.jssip.net/js/custom.js) 404 (Not Found) [tryit.jssip.net/:22](http://tryit.jssip.net/:22) event.returnValue is deprecated. Please use the standard event.preventDefault() instead. jquery-1.6.1.min.js:17 JsSIP | UA | configuration parameters after validation: jssip-devel.js:5783 • via\_host: “7h1b1f5fc8gi.invalid” jssip-devel.js:5794 • password: NOT SHOWN jssip-devel.js:5791 • register\_expires: 600 jssip-devel.js:5794 • register\_min\_expires: 120 jssip-devel.js:5794 • register: true jssip-devel.js:5794 • registrar\_server: sip:webrtc.no-ip.org jssip-devel.js:5788 • ws\_server\_max\_reconnection: 3 jssip-devel.js:5794 • ws\_server\_reconnection\_timeout: 4 jssip-devel.js:5794 • connection\_recovery\_min\_interval: 2 jssip-devel.js:5794 • connection\_recovery\_max\_interval: 30 jssip-devel.js:5794 • use\_preloaded\_route: false jssip-devel.js:5794 • no\_answer\_timeout: 60000 jssip-devel.js:5794 • stun\_servers: [“stun:74.125.132.127:19302”] jssip-devel.js:5794 • turn\_servers: [] jssip-devel.js:5794 • trace\_sip: true jssip-devel.js:5794 • hack\_via\_tcp: false jssip-devel.js:5794 • hack\_ip\_in\_contact: false jssip-devel.js:5794 • uri: [sip:8001@webrtc.no-ip.org](mailto:sip:8001@webrtc.no-ip.org) jssip-devel.js:5788 • ws\_servers: [{“ws\_uri”:“ws://webrtc.no-ip.org:8088/ws”,“sip\_uri”:“sip:webrtc.no-ip.org:8088;transport=ws;lr”,“weight”:0,“status”:0,“scheme”:“WS”}] jssip-devel.js:5794 • display\_name: “G” jssip-devel.js:5794 • instance\_id: “1b399baa-f209-4868-9e1e-490fb50fb0ea” jssip-devel.js:5794 • jssip\_id: “cgu3s” jssip-devel.js:5794 • hostport\_params: “[webrtc.no-ip.org](http://webrtc.no-ip.org)” jssip-devel.js:5794 • authorization\_user: “8001” jssip-devel.js:5794 JsSIP | EVENT EMITTER | adding event newMessage jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event newRTCSession jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event registrationFailed jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event unregistered jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event registered jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event disconnected jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event connected jssip-devel.js:67 JsSIP | EVENT EMITTER | new listener added to event connected jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event disconnected jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event newRTCSession jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event newMessage jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event registered jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event unregistered jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event registrationFailed jssip-devel.js:63 JsSIP | UA | user requested startup… jssip-devel.js:5238 JsSIP | TRANSPORT | connecting to WebSocket ws://webrtc.no-ip.org:8088/ws jssip-devel.js:568 JsSIP | TRANSPORT | WebSocket ws://webrtc.no-ip.org:8088/ws connected jssip-devel.js:604 JsSIP | UA | connection state set to 0 jssip-devel.js:5360 JsSIP | EVENT EMITTER | emitting event connected jssip-devel.js:187 JsSIP | TRANSPORT | sending WebSocket message: REGISTER sip:webrtc.no-ip.org SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK4913074 Max-Forwards: 69 To: sip:8001@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci Call-ID: m94e59t1sqbl2pobami4no CSeq: 81 REGISTER Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:1b399baa-f209-4868-9e1e-490fb50fb0ea”;expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 jssip-devel.js:519 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK4913074;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci To: sip:8001@webrtc.no-ip.org;tag=as320bbfd8 Call-ID: m94e59t1sqbl2pobami4no CSeq: 81 REGISTER Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce="15ee906d" Content-Length: 0 jssip-devel.js:686 JsSIP | TRANSPORT | sending WebSocket message: REGISTER sip:webrtc.no-ip.org SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK8025886 Max-Forwards: 69 To: sip:8001@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci Call-ID: m94e59t1sqbl2pobami4no CSeq: 82 REGISTER Authorization: Digest algorithm=MD5, username=“8001”, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce=“15ee906d”, uri=“sip:webrtc.no-ip.org”, response="332f925df91f25dbfbaf82267a6c87d1" Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;reg-id=1;+sip.instance=“urn:uuid:1b399baa-f209-4868-9e1e-490fb50fb0ea”;expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 jssip-devel.js:519 JsSIP | TRANSPORT | received WebSocket text message: OPTIONS sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws SIP/2.0 Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK56a57557 Max-Forwards: 70 From: “asterisk” sip:8001@192.168.192.7;tag=as37fa2738 To: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws Contact: sip:8001@192.168.192.7:5060;transport=WS Call-ID: 63c9aa52286750c914d6d89250525a51@192.168.192.7:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.7.0 Date: Fri, 25 Apr 2014 01:44:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 jssip-devel.js:686 JsSIP | TRANSPORT | sending WebSocket message: SIP/2.0 200 OK Via: SIP/2.0/WS 192.168.192.7:5060;branch=z9hG4bK56a57557 To: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;tag=nljv5atrao From: “asterisk” sip:8001@192.168.192.7;tag=as37fa2738 Call-ID: 63c9aa52286750c914d6d89250525a51@192.168.192.7:5060 CSeq: 102 OPTIONS Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Accept: application/sdp,application/dtmf-relay Content-Length: 0 jssip-devel.js:519 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK8025886;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=lftr2g6uci To: sip:8001@webrtc.no-ip.org;tag=as320bbfd8 Call-ID: m94e59t1sqbl2pobami4no CSeq: 82 REGISTER Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Expires: 600 Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;expires=600 Date: Fri, 25 Apr 2014 01:44:11 GMT Content-Length: 0 jssip-devel.js:686 JsSIP | EVENT EMITTER | emitting event registered jssip-devel.js:187 JsSIP | TRANSACTION | Timer J expired for non-INVITE server transaction z9hG4bK56a57557 jssip-devel.js:2094 Registered init.js:428 JsSIP | EVENT EMITTER | adding event newDTMF jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event ended jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event started jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event failed jssip-devel.js:67 JsSIP | EVENT EMITTER | adding event progress jssip-devel.js:67 JsSIP | EVENT EMITTER | emitting event newRTCSession jssip-devel.js:187 JsSIP | RTC SESSION | requesting access to local media jssip-devel.js:3442 JsSIP | EVENT EMITTER | new listener added to event progress jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event started jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event failed jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event newDTMF jssip-devel.js:63 JsSIP | EVENT EMITTER | new listener added to event ended jssip-devel.js:63 JsSIP | RTC SESSION | got local media stream jssip-devel.js:3446 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1117122343 1 udp 2122260223 192.168.101.103 54280 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1117122343 2 udp 2122260223 192.168.101.103 54280 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1755041049 1 udp 2122194687 192.168.74.1 54281 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:1755041049 2 udp 2122194687 192.168.74.1 54281 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:4254299990 1 udp 2122129151 10.246.40.1 54282 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:4254299990 2 udp 2122129151 10.246.40.1 54282 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0 jssip-devel.js:3401 JsSIP | RTC SESSION | ICE candidate received: a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0 jssip-devel.js:3401 JsSIP | TRANSPORT | sending WebSocket message: INVITE [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv Call-ID: cgu3s60etphsvr40mneg CSeq: 5919 INVITE Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;ob Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 2242 v=0 o=- 3652154159910496246 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS 5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 m=audio 54280 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.101.103 a=rtcp:54280 IN IP4 192.168.101.103 a=candidate:1117122343 1 udp 2122260223 192.168.101.103 54280 typ host generation 0 a=candidate:1117122343 2 udp 2122260223 192.168.101.103 54280 typ host generation 0 a=candidate:1755041049 1 udp 2122194687 192.168.74.1 54281 typ host generation 0 a=candidate:1755041049 2 udp 2122194687 192.168.74.1 54281 typ host generation 0 a=candidate:4254299990 1 udp 2122129151 10.246.40.1 54282 typ host generation 0 a=candidate:4254299990 2 udp 2122129151 10.246.40.1 54282 typ host generation 0 a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=ice-ufrag:eNWxE4gTTqnCVyMu a=ice-pwd:ZoVKNOv2HlfHE9VNyW3NSIAO a=ice-options:google-ice a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:tCMlVm6sT9MJMxbzKEmsDV1e3VVvNa5MwkUKPKzN a=crypto:1 AES\_CM\_128\_HMAC\_SHA1\_80 inline:qHzInMoGik8HNFMVgi6IjHdClreeyslHlw11nvn8 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 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:3951139034 cname:1GEOwdZBp2a0BbWt a=ssrc:3951139034 msid:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 9cda3eb5-5a5c-4cb3-ad6d-0a153309d435 a=ssrc:3951139034 mslabel:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 a=ssrc:3951139034 label:9cda3eb5-5a5c-4cb3-ad6d-0a153309d435 jssip-devel.js:519 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 401 Unauthorized Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv To: sip:1000@webrtc.no-ip.org;tag=as0464893c Call-ID: cgu3s60etphsvr40mneg CSeq: 5919 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce="12671f92" Content-Length: 0 jssip-devel.js:686 JsSIP | TRANSPORT | sending WebSocket message: ACK [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK9976051 To: sip:1000@webrtc.no-ip.org;tag=as0464893c From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv Call-ID: cgu3s60etphsvr40mneg CSeq: 5919 ACK jssip-devel.js:519 JsSIP | TRANSPORT | sending WebSocket message: INVITE [sip:1000@webrtc.no-ip.org](mailto:sip:1000@webrtc.no-ip.org) SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv Call-ID: cgu3s60etphsvr40mneg CSeq: 5920 INVITE Authorization: Digest algorithm=MD5, username=“8001”, realm=“[webrtc.no-ip.org](http://webrtc.no-ip.org)”, nonce=“12671f92”, [uri="sip:1000@webrtc.no-ip.org](mailto:uri=%22sip:1000@webrtc.no-ip.org)", response="63ef359a7befb51ebef6a306a7262d82" Contact: sip:06sfurc9@7h1b1f5fc8gi.invalid;transport=ws;ob Allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE Content-Type: application/sdp Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 2242 v=0 o=- 3652154159910496246 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS 5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 m=audio 54280 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 192.168.101.103 a=rtcp:54280 IN IP4 192.168.101.103 a=candidate:1117122343 1 udp 2122260223 192.168.101.103 54280 typ host generation 0 a=candidate:1117122343 2 udp 2122260223 192.168.101.103 54280 typ host generation 0 a=candidate:1755041049 1 udp 2122194687 192.168.74.1 54281 typ host generation 0 a=candidate:1755041049 2 udp 2122194687 192.168.74.1 54281 typ host generation 0 a=candidate:4254299990 1 udp 2122129151 10.246.40.1 54282 typ host generation 0 a=candidate:4254299990 2 udp 2122129151 10.246.40.1 54282 typ host generation 0 a=candidate:202773463 1 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:202773463 2 tcp 1518280447 192.168.101.103 0 typ host generation 0 a=candidate:639119849 1 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:639119849 2 tcp 1518214911 192.168.74.1 0 typ host generation 0 a=candidate:3004205990 1 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=candidate:3004205990 2 tcp 1518149375 10.246.40.1 0 typ host generation 0 a=ice-ufrag:eNWxE4gTTqnCVyMu a=ice-pwd:ZoVKNOv2HlfHE9VNyW3NSIAO a=ice-options:google-ice a=fingerprint:sha-256 36:BB:11:6A:0A:2B:56:26:D5:E0:1F:98:8A:A0:4E:EF:93:0A:3A:D3:84:A5:EF:14:C6:C8:86:D1:AF:1A:08:84 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=sendrecv a=rtcp-mux a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:tCMlVm6sT9MJMxbzKEmsDV1e3VVvNa5MwkUKPKzN a=crypto:1 AES\_CM\_128\_HMAC\_SHA1\_80 inline:qHzInMoGik8HNFMVgi6IjHdClreeyslHlw11nvn8 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 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:3951139034 cname:1GEOwdZBp2a0BbWt a=ssrc:3951139034 msid:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 9cda3eb5-5a5c-4cb3-ad6d-0a153309d435 a=ssrc:3951139034 mslabel:5RwvfIQD4tReS2G5yvV1x6j62I2NOFA1nO36 a=ssrc:3951139034 label:9cda3eb5-5a5c-4cb3-ad6d-0a153309d435 jssip-devel.js:519 JsSIP | TRANSACTION | Timer D expired for INVITE client transaction z9hG4bK9976051 jssip-devel.js:1969 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 100 Trying Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv To: sip:1000@webrtc.no-ip.org Call-ID: cgu3s60etphsvr40mneg CSeq: 5920 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:1000@192.168.192.7:5060;transport=WS Content-Length: 0 jssip-devel.js:686 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK7110183;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8 Call-ID: cgu3s60etphsvr40mneg CSeq: 5920 INVITE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:1000@192.168.192.7:5060;transport=WS Content-Type: application/sdp Content-Length: 827 v=0 o=root 1916711608 1916711608 IN IP4 192.168.192.7 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.192.7 t=0 0 m=audio 16094 RTP/SAVPF 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:508b3b3a55b6364115e87667717fe8b3 a=ice-pwd:12147891493792b1612c66d02e20cfd0 a=candidate:Hc0a8c007 1 UDP 2130706431 192.168.192.7 16094 typ host a=candidate:Hc0a80103 1 UDP 2130706431 192.168.1.3 16094 typ host a=candidate:Sc953b8ef 1 UDP 1694498815 201.83.184.239 10332 typ srflx a=candidate:Hc0a8c007 2 UDP 2130706430 192.168.192.7 16095 typ host a=candidate:Hc0a80103 2 UDP 2130706430 192.168.1.3 16095 typ host a=candidate:Sc953b8ef 2 UDP 1694498814 201.83.184.239 10332 typ srflx a=sendrecv a=crypto:0 AES\_CM\_128\_HMAC\_SHA1\_32 inline:5HrqW3GMP29G3i/zeH2c6B1DTbOS0UNdeeuY6i/f jssip-devel.js:686 JsSIP | DIALOG | new UAC dialog created with status CONFIRMED jssip-devel.js:2546 JsSIP | RTC SESSION | stream added: default jssip-devel.js:3392 JsSIP | TRANSPORT | sending WebSocket message: ACK sip:1000@192.168.192.7:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3660144 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv Call-ID: cgu3s60etphsvr40mneg CSeq: 5920 ACK Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 jssip-devel.js:519 JsSIP | EVENT EMITTER | emitting event started jssip-devel.js:187 JsSIP | TRANSACTION | Timer B expired for INVITE client transaction z9hG4bK7110183 jssip-devel.js:1960 JsSIP | TRANSACTION | Timer M expired for INVITE client transaction z9hG4bK7110183 jssip-devel.js:1949 JsSIP | RTC SESSION | terminating RTCSession jssip-devel.js:3790 JsSIP | TRANSPORT | sending WebSocket message: BYE sip:1000@192.168.192.7:5060;transport=ws SIP/2.0 Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3459967 Max-Forwards: 69 To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv Call-ID: cgu3s60etphsvr40mneg CSeq: 5921 BYE Supported: path, outbound, gruu User-Agent: JsSIP 0.3.7 Content-Length: 0 jssip-devel.js:519 JsSIP | RTC SESSION | closing INVITE session cgu3s60etphsvr40mnegi6i4nniqbv jssip-devel.js:4248 JsSIP | RTC SESSION | closing PeerConnection jssip-devel.js:3424 JsSIP | DIALOG | dialog cgu3s60etphsvr40mnegi6i4nniqbvas40e79ab8 deleted jssip-devel.js:2566 JsSIP | EVENT EMITTER | emitting event ended jssip-devel.js:187 JsSIP | TRANSPORT | received WebSocket text message: SIP/2.0 200 OK Via: SIP/2.0/WS 7h1b1f5fc8gi.invalid;branch=z9hG4bK3459967;received=189.232.221.54 From: “G” sip:8001@webrtc.no-ip.org;tag=i6i4nniqbv To: sip:1000@webrtc.no-ip.org;tag=as40e79ab8 Call-ID: cgu3s60etphsvr40mneg CSeq: 5921 BYE Server: Asterisk PBX 11.7.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 [/code] ---
**Author:** ![navaismo](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/navaismo/32/2481_2.png) [@navaismo](https://community.asterisk.org/u/navaismo)\ **Post date:** [April 25, 2014, 12:15pm UTC](https://community.asterisk.org/t/webrtc-sound-problems/45983/6 "2014-04-25T12:15:50Z")
The SDP is using a diferent IP, the signaling send the INVITE from 179.213.21.254 and the SDP use: 192.168.101.103. This issue is related to your NAT settings or ice/stun servers. ---
**Author:** ![gfesj](https://avatars.discourse-cdn.com/v4/letter/g/a4c791/32.png) [@gfesj](https://community.asterisk.org/u/gfesj)\ **Post date:** [April 25, 2014, 3:29pm UTC](https://community.asterisk.org/t/webrtc-sound-problems/45983/7 "2014-04-25T15:29:02Z")
Thank you very much! In my sip.conf i used nat=auto\_force\_rport,auto\_comedia I just alter to nat=force\_rport,comedia and the audio start to work. Thanks Geraldo