<div dir="ltr"><div><div><div><div><div><div>Thanks. <br><br></div>I almost have things working (same chrome as you are using).<br><br></div>I'm getting a chrome issue, OS Ubuntu 14.04. "Denied Media Access". <br><br></div>If I'm using chrome under OSX (<a href="http://tryit.jssip.net">tryit.jssip.net</a>) to someone jitsi, seems to have worked briefly (just not tested throughly yet).<br><br></div>Dont' think this has anything to do with spce, but would appreciate anyones thoughts of chrome under ubuntu (googling seems to show this issue as well on other non spce things, but don't know the work around).<br><br></div>Yours truly,<br></div>Brian<br><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Jul 9, 2015 at 12:01 AM, Daniel Grotti <span dir="ltr"><<a href="mailto:dgrotti@sipwise.com" target="_blank">dgrotti@sipwise.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Brian,<br>
this is a working configuration for Chrome Version 43.0.2357.65:<br>
<br>
<br>
ws://yourip:5060/ws, wss://yourip:5061/ws, wss://yourip:1443/wss/sip/<br>
<br>
<br>
* WebRTC Subscribers -> Details -> Preferences -> NAT and Media Flow Control<br>
<br>
- 'use_rtpproxy:' Always with rtpproxy as additional/only ICE candidate<br>
- 'transport_protocol:' RTP/SAVPF (encrypted SRTP with RTCP feedback) <br>
(for Chrome Version 43.0.2357.65)<br>
<br>
* Domain -> Details -> Preferences -> NAT and Media Flow Control<br>
<br>
- 'transport_protocol:' RTP/AVP (Plain RTP)<br>
<br>
<br>
--<br>
Daniel Grotti<br>
VoIP Engineer<br>
<br>
<br>
Sipwise GmbH<br>
Europaring F15 | 2345 Brunn am Gebirge, Austria | <a href="http://www.sipwise.com" rel="noreferrer" target="_blank">www.sipwise.com</a><br>
<br>
On 07/09/2015 01:30 AM, Brian Quandt wrote:<br>
> What does this mean in the rtp.log?<br>
><br>
> I've not yet been able to get jitsi or jssip to work completely. But I<br>
> can get users to login/register on each clietn, and they can call each<br>
> other (but upon asnwering, nothing, or a webrtc failure in jssip).<br>
><br>
> Maybe the "unknown codec" is the issue?<br>
><br>
> Yours truly,<br>
> Brian<br>
><br>
><br>
> A89F8-D5A5F700'<br>
> Jul 8 23:23:45 ip-10-220-196-230 rtpengine[6430]:<br>
> [a9e2e70e7cb3bb435b081bc5e2469ee2@0:0:0:0:0:0:0:0] ------ Media #1<br>
> (audio over UDP/TLS/RTP/SAVPF) using unknown codec<br>
> Jul 8 23:23:45 ip-10-220-196-230 rtpengine[6430]:<br>
> [a9e2e70e7cb3bb435b081bc5e2469ee2@0:0:0:0:0:0:0:0] --------- Port 32832<br>
> <> <a href="http://157.254.210.17:5000" rel="noreferrer" target="_blank">157.254.210.17:5000</a> <<a href="http://157.254.210.17:5000" rel="noreferrer" target="_blank">http://157.254.210.17:5000</a>> , 0 p, 0 b, 0 e,<br>
> 1436397570 last_packet<br>
> Jul 8 23:23:45 ip-10-220-196-230 rtpengine[6430]:<br>
> [a9e2e70e7cb3bb435b081bc5e2469ee2@0:0:0:0:0:0:0:0] --------- Port 32833<br>
> <> <a href="http://157.254.210.17:5001" rel="noreferrer" target="_blank">157.254.210.17:5001</a> <<a href="http://157.254.210.17:5001" rel="noreferrer" target="_blank">http://157.254.210.17:5001</a>> (RTCP), 0 p, 0 b,<br>
> 0 e, 1436397570 last_packet<br>
> Jul 8 23:23:45 ip-10-220-196-230 rtpengine[6430]:<br>
> [a9e2e70e7cb3bb435b081bc5e2469ee2@0:0:0:0:0:0:0:0] ------ Media #2<br>
> (video over UDP/TLS/RTP/SAVPF) using unknown codec<br>
> Jul 8 23:23:45 ip-10-220-196-230 rtpengine[6430]:<br>
> [a9e2e70e7cb3bb435b081bc5e2469ee2@0:0:0:0:0:0:0:0] --------- Port 32846<br>
> <> <a href="http://157.254.210.17:5016" rel="noreferrer" target="_blank">157.254.210.17:5016</a> <<a href="http://157.254.210.17:5016" rel="noreferrer" target="_blank">http://157.254.210.17:5016</a>> , 0 p, 0 b, 0 e,<br>
> 1436397570 last_packet<br>
><br>
><br>
> On Wed, Jul 8, 2015 at 3:03 PM, Brian Quandt <<a href="mailto:brian.quandt@gmail.com">brian.quandt@gmail.com</a><br>
> <mailto:<a href="mailto:brian.quandt@gmail.com">brian.quandt@gmail.com</a>>> wrote:<br>
><br>
> Daniel,<br>
><br>
> Thanks for the reply. It got me looking, but the instructions as<br>
> references actually was a step backwards. Lost all connectivity for<br>
> all users, plus the mods to the user I'd set for webrtc couldn't<br>
> even connect/register per jitsi.<br>
><br>
> Restoring everything back to it's original I'm back again, with<br>
> users able to connect to one another, and the failured in webrtc...<br>
><br>
> But as I said you have me hopefully looking in the right direction.<br>
><br>
> Yours truly,<br>
> Brian<br>
><br>
><br>
> On Wed, Jul 8, 2015 at 1:47 PM, Daniel Grotti <<a href="mailto:dgrotti@sipwise.com">dgrotti@sipwise.com</a><br>
> <mailto:<a href="mailto:dgrotti@sipwise.com">dgrotti@sipwise.com</a>>> wrote:<br>
><br>
> Hi Brian,<br>
> Maybe this could be a good start:<br>
> <a href="https://www.linkedin.com/pulse/how-enable-webrtc-sipprovider-daniel-grotti?_mSplash=1" rel="noreferrer" target="_blank">https://www.linkedin.com/pulse/how-enable-webrtc-sipprovider-daniel-grotti?_mSplash=1</a><br>
><br>
> Please notice that you may need to configure the<br>
> transport_protocol in ngcp toeard s the webrtc client in a<br>
> different way. Depends on the browser you are using.<br>
><br>
> Daniel<br>
><br>
> On Jul 8, 2015 9:19 PM, Brian Quandt <<a href="mailto:brian.quandt@gmail.com">brian.quandt@gmail.com</a><br>
> <mailto:<a href="mailto:brian.quandt@gmail.com">brian.quandt@gmail.com</a>>> wrote:<br>
> ><br>
> > Trying to get things working and am stumbling. Maybe someone<br>
> can help me a bit?<br>
> ><br>
> > Right now, I just want to get things working, ie do a simple<br>
> test using <a href="http://jssip.net" rel="noreferrer" target="_blank">jssip.net</a> <<a href="http://jssip.net" rel="noreferrer" target="_blank">http://jssip.net</a>>, based on the AWS AMI<br>
> image built by sipwise, ie sip:provider CE AMI mr3.8.2, image<br>
> id: ami-17142e27 (us west 2)<br>
> ><br>
> > Here's my steps so far:<br>
> ><br>
> > 1) got the ec2 instance running<br>
> > 2) configured the ec2 security group/ports as below:<br>
> ><br>
> > HTTP<br>
> > TCP<br>
> > 80<br>
> > <a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">0.0.0.0/0</a> <<a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">http://0.0.0.0/0</a>><br>
> > HTTPS<br>
> > TCP<br>
> > 443<br>
> > <a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">0.0.0.0/0</a> <<a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">http://0.0.0.0/0</a>><br>
> > Custom TCP Rule<br>
> > TCP<br>
> > 1080<br>
> > <a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">0.0.0.0/0</a> <<a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">http://0.0.0.0/0</a>><br>
> > Custom TCP Rule<br>
> > TCP<br>
> > 1443<br>
> > <a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">0.0.0.0/0</a> <<a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">http://0.0.0.0/0</a>><br>
> > Custom TCP Rule<br>
> > TCP<br>
> > 2443<br>
> > <a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">0.0.0.0/0</a> <<a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">http://0.0.0.0/0</a>><br>
> > Custom TCP Rule<br>
> > TCP<br>
> > 5060<br>
> > <a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">0.0.0.0/0</a> <<a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">http://0.0.0.0/0</a>><br>
> > Custom TCP Rule<br>
> > TCP<br>
> > 5061<br>
> > <a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">0.0.0.0/0</a> <<a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">http://0.0.0.0/0</a>><br>
> > Custom UDP Rule<br>
> > UDP<br>
> > 5060<br>
> > <a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">0.0.0.0/0</a> <<a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">http://0.0.0.0/0</a>><br>
> > Custom UDP Rule<br>
> > UDP<br>
> > 5061<br>
> > <a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">0.0.0.0/0</a> <<a href="http://0.0.0.0/0" rel="noreferrer" target="_blank">http://0.0.0.0/0</a>><br>
> > ssh is configured for my machine only (obviously)<br>
> ><br>
> > 3) got a proper ssl cert from godaddy, change all my<br>
> sslcerfile and sslkey files in config.yml appropriately, and<br>
> made sure kamailio tls is enabled (which it is by default in the<br>
> ami) ran ngcpcfg apply (everything was happy so far).<br>
> ><br>
> > 4) launched firefox under linux going to <a href="http://tryit.jssip.net" rel="noreferrer" target="_blank">tryit.jssip.net</a><br>
> <<a href="http://tryit.jssip.net" rel="noreferrer" target="_blank">http://tryit.jssip.net</a>>, with folowing details:<br>
> > name: quandt<br>
> > sip uri: <a href="mailto:sip%3Aquandt@sip.autodcp.com">sip:quandt@sip.autodcp.com</a><br>
> <mailto:<a href="mailto:sip%253Aquandt@sip.autodcp.com">sip%3Aquandt@sip.autodcp.com</a>><br>
> > password: ******<br>
> > ws uri: wss://<a href="http://sip.autodcp.com:1443/wss/sip/" rel="noreferrer" target="_blank">sip.autodcp.com:1443/wss/sip/</a><br>
> <<a href="http://sip.autodcp.com:1443/wss/sip/" rel="noreferrer" target="_blank">http://sip.autodcp.com:1443/wss/sip/</a>><br>
> ><br>
> > Which got me to the jssip demo page both connected and<br>
> registered just fine.<br>
> ><br>
> > 5) on a mac launched zoiper and logged into another account on<br>
> my sip server<br>
> ><br>
> > 6) tried to call from one to the other. Got a ring from one<br>
> ot the other to work, on the jssip demo page, when I ansewred, I<br>
> get promoted to share my microphone, which I acknowlege, and<br>
> them get a WebRTC error right away. Below is part of the<br>
> console messages.<br>
> ><br>
> > Any thoughts?<br>
> ><br>
> > Yours truly,<br>
> > Brian<br>
> ><br>
> ><br>
> ><br>
> > " " +2s" jssip.js:21621<br>
> > "JsSIP:Transport " "sending WebSocket message:<br>
> ><br>
> > SIP/2.0 200 OK<br>
> ><br>
> > Via: SIP/2.0/WSS<br>
> 54.189.6.185:5061;branch=z9hG4bKdf8e.4e2788dc9d9e3774bc01623f84011d82.0<br>
> ><br>
> > Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0<br>
> ><br>
> > To: sip:d62a2g56@ug5tmpr4sfhc.invalid;transport=ws;tag=je09o4s8o3<br>
> ><br>
> > From:<br>
> sip:pinger@sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-8205b6a6<br>
> ><br>
> > Call-ID: <a href="mailto:8c334a51-c57f0854-1af23e2@127.0.0.1">8c334a51-c57f0854-1af23e2@127.0.0.1</a><br>
> <mailto:<a href="mailto:8c334a51-c57f0854-1af23e2@127.0.0.1">8c334a51-c57f0854-1af23e2@127.0.0.1</a>><br>
> ><br>
> > CSeq: 1 OPTIONS<br>
> ><br>
> > Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS<br>
> ><br>
> > Accept: application/sdp, application/dtmf-relay<br>
> ><br>
> > Supported: outbound<br>
> ><br>
> > Content-Length: 0<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > " " +16ms" jssip.js:21621<br>
> > "JsSIP:NonInviteServerTransaction " "Timer J expired for<br>
> transaction z9hG4bKdf8e.4e2788dc9d9e3774bc01623f84011d82.0" "<br>
> +3ms" jssip.js:21621<br>
> > "JsSIP:RTCSession " "answer()" " +1s" jssip.js:21621<br>
> > "JsSIP:Dialog " "dialog<br>
> MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-12mftbnm4nm5A6A45EC-559D727B0008760D-D4D52700<br>
> changed to CONFIRMED state" " +0ms" jssip.js:21621<br>
> > "rtcninja:RTCPeerConnection " "new | pcConfig: " Object {<br>
> iceServers: Array[1], gatheringTimeout: 2000 } " +3ms"<br>
> jssip.js:21621<br>
> > "rtcninja:RTCPeerConnection " "setConfigurationAndOptions |<br>
> processed pcConfig: " Object { iceServers: Array[1] } " +1ms"<br>
> jssip.js:21621<br>
> > "rtcninja:Adapter " "getUserMedia() | constraints: " Object {<br>
> audio: true, video: false } " +93ms" jssip.js:21621<br>
> > Invalid URI. Load of media resource failed. <a href="http://tryit.jssip.net" rel="noreferrer" target="_blank">tryit.jssip.net</a><br>
> <<a href="http://tryit.jssip.net" rel="noreferrer" target="_blank">http://tryit.jssip.net</a>><br>
> > "rtcninja:Adapter " "getUserMedia() | success" " +2s"<br>
> jssip.js:21621<br>
> > "rtcninja:RTCPeerConnection " "addStream() | stream: [object<br>
> LocalMediaStream]" " +0ms" jssip.js:21621<br>
> > "rtcninja:RTCPeerConnection " "setRemoteDescription()" " +1ms"<br>
> jssip.js:21621<br>
> > "rtcninja:ERROR:RTCPeerConnection " "setRemoteDescription() |<br>
> error:" " +1ms" Object { name: "INVALID_SESSION_DESCRIPTION",<br>
> message: "Could not negotiate media lines; cause =<br>
> NO_DTLS_FINGERPRINT | SDP Parsing Error: Warning: No network<br>
> type specified in comediadir attribute.", __exposedProps__:<br>
> Object } jssip.js:21796<br>
> > "JsSIP:Transport " "sending WebSocket message:<br>
> ><br>
> > SIP/2.0 488 Not Acceptable Here<br>
> ><br>
> > Via: SIP/2.0/WSS<br>
> 54.189.6.185:5061;branch=z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0<br>
> ><br>
> > Via: SIP/2.0/UDP<br>
> 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKuN~pJa9J;rport=5080<br>
> ><br>
> > To: <<a href="mailto:sip%3Ad62a2g56@sip.autodcp.com">sip:d62a2g56@sip.autodcp.com</a><br>
> <mailto:<a href="mailto:sip%253Ad62a2g56@sip.autodcp.com">sip%3Ad62a2g56@sip.autodcp.com</a>>>;tag=2mftbnm4nm<br>
> ><br>
> > From: <<a href="mailto:sip%3A0991002@sip.autodcp.com">sip:0991002@sip.autodcp.com</a><br>
> <mailto:<a href="mailto:sip%253A0991002@sip.autodcp.com">sip%3A0991002@sip.autodcp.com</a>>>;tag=5A6A45EC-559D727B0008760D-D4D52700<br>
> ><br>
> > Call-ID: MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-1<br>
> ><br>
> > CSeq: 10 INVITE<br>
> ><br>
> > Supported: timer,ice,outbound<br>
> ><br>
> > Content-Length: 0<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > " " +0ms" jssip.js:21621<br>
> > "JsSIP:RTCSession " "session failed" " +1ms" jssip.js:21621<br>
> > "JsSIP:RTCSession " "close()" " +0ms" jssip.js:21621<br>
> > "rtcninja:RTCPeerConnection " "close()" " +0ms" jssip.js:21621<br>
> > "rtcninja:RTCPeerConnection " "oniceconnectionstatechange() |<br>
> iceConnectionState: closed" " +0ms" jssip.js:21621<br>
> > "JsSIP:RTCSession " "close() | closing local MediaStream" "<br>
> +0ms" jssip.js:21621<br>
> > "rtcninja:Adapter " "closeMediaStream() | calling stop() on<br>
> all the MediaStreamTrack" " +1ms" jssip.js:21621<br>
> > "JsSIP:Dialog " "dialog<br>
> MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-12mftbnm4nm5A6A45EC-559D727B0008760D-D4D52700<br>
> deleted" " +4ms" jssip.js:21621<br>
> > "rtcninja:RTCPeerConnection " "onsignalingstatechange() |<br>
> signalingState: closed" " +5ms" jssip.js:21621<br>
> > "JsSIP:Transport " "received WebSocket text message:<br>
> ><br>
> > ACK sip:d62a2g56@ug5tmpr4sfhc.invalid;transport=ws SIP/2.0<br>
> ><br>
> > Max-Forwards: 70<br>
> ><br>
> > Record-Route:<br>
> <sip:54.189.6.185:5060;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061;ftag=5A6A45EC-559D727B0008760D-D4D52700;lr=on><br>
> ><br>
> > Record-Route:<br>
> <sip:127.0.0.1:5060;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061;ftag=5A6A45EC-559D727B0008760D-D4D52700;lr=on><br>
> ><br>
> > Via: SIP/2.0/WSS<br>
> 54.189.6.185:5061;branch=z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0<br>
> ><br>
> > Via: SIP/2.0/UDP<br>
> 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKuN~pJa9J;rport=5080<br>
> ><br>
> > From: <<a href="mailto:sip%3A0991002@sip.autodcp.com">sip:0991002@sip.autodcp.com</a><br>
> <mailto:<a href="mailto:sip%253A0991002@sip.autodcp.com">sip%3A0991002@sip.autodcp.com</a>>>;tag=5A6A45EC-559D727B0008760D-D4D52700<br>
> ><br>
> > To: <<a href="mailto:sip%3Ad62a2g56@sip.autodcp.com">sip:d62a2g56@sip.autodcp.com</a><br>
> <mailto:<a href="mailto:sip%253Ad62a2g56@sip.autodcp.com">sip%3Ad62a2g56@sip.autodcp.com</a>>>;tag=2mftbnm4nm<br>
> ><br>
> > Call-ID: MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-1<br>
> ><br>
> > CSeq: 10 ACK<br>
> ><br>
> > Content-Length: 0<br>
> ><br>
> > Route: <sip:10.220.196.230:32769;transport=ws><br>
> ><br>
> ><br>
> ><br>
> ><br>
> > " " +31ms" jssip.js:21621<br>
> > "JsSIP:Transport " "received WebSocket text message:<br>
> ><br>
> > OPTIONS sip:d62a2g56@ug5tmpr4sfhc.invalid;transport=ws SIP/2.0<br>
> ><br>
> > Max-Forwards: 70<br>
> ><br>
> > Record-Route:<br>
> <sip:54.189.6.185:5060;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061;ftag=uloc-559adce7-19bb-91-ac5e6b73-c205b6a6;lr=on><br>
> ><br>
> > Record-Route:<br>
> <sip:127.0.0.1:5060;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061;ftag=uloc-559adce7-19bb-91-ac5e6b73-c205b6a6;lr=on><br>
> ><br>
> > Via: SIP/2.0/WSS<br>
> 54.189.6.185:5061;branch=z9hG4bK8d0f.6a6365b82f7b9a11de527fd7f0d652bd.0<br>
> ><br>
> > Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0<br>
> ><br>
> > From:<br>
> sip:pinger@sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-c205b6a6<br>
> ><br>
> > To: sip:d62a2g56@ug5tmpr4sfhc.invalid;transport=ws<br>
> ><br>
> > Call-ID: <a href="mailto:8c334a51-067f0854-fbf23e2@127.0.0.1">8c334a51-067f0854-fbf23e2@127.0.0.1</a><br>
> <mailto:<a href="mailto:8c334a51-067f0854-fbf23e2@127.0.0.1">8c334a51-067f0854-fbf23e2@127.0.0.1</a>><br>
> ><br>
> > CSeq: 1 OPTIONS<br>
> ><br>
> > Content-Length: 0<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > " " +27s" jssip.js:21621<br>
> > "JsSIP:Transport " "sending WebSocket message:<br>
> ><br>
> > SIP/2.0 200 OK<br>
> ><br>
> > Via: SIP/2.0/WSS<br>
> 54.189.6.185:5061;branch=z9hG4bK8d0f.6a6365b82f7b9a11de527fd7f0d652bd.0<br>
> ><br>
> > Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0<br>
> ><br>
> > To: sip:d62a2g56@ug5tmpr4sfhc.invalid;transport=ws;tag=5r8pi0ggps<br>
> ><br>
> > From:<br>
> sip:pinger@sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-c205b6a6<br>
> ><br>
> > Call-ID: <a href="mailto:8c334a51-067f0854-fbf23e2@127.0.0.1">8c334a51-067f0854-fbf23e2@127.0.0.1</a><br>
> <mailto:<a href="mailto:8c334a51-067f0854-fbf23e2@127.0.0.1">8c334a51-067f0854-fbf23e2@127.0.0.1</a>><br>
> ><br>
> > CSeq: 1 OPTIONS<br>
> ><br>
> > Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS<br>
> ><br>
> > Accept: application/sdp, application/dtmf-relay<br>
> ><br>
> > Supported: outbound<br>
> ><br>
> > Content-Length: 0<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > " " +7ms" jssip.js:21621<br>
> > "JsSIP:NonInviteServerTransaction " "Timer J expired for<br>
> transaction z9hG4bK8d0f.6a6365b82f7b9a11de527fd7f0d652bd.0" "<br>
> +0ms" jssip.js:21621<br>
> > "JsSIP:InviteServerTransaction " "Timer H expired for<br>
> transaction z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0" "<br>
> +5s" jssip.js:21621<br>
> > "JsSIP:Transport " "received WebSocket text message:<br>
> ><br>
> > OPTIONS sip:d62a2g56@ug5tmpr4sfhc.invalid;transport=ws SIP/2.0<br>
> ><br>
> > Max-Forwards: 70<br>
> ><br>
> > Record-Route:<br>
> <sip:54.189.6.185:5060;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061;ftag=uloc-559adce7-19bb-91-ac5e6b73-0305b6a6;lr=on><br>
> ><br>
> > Record-Route:<br>
> <sip:127.0.0.1:5060;ngcplb=yes;r2=on;socket=sip:10.220.196.230:5061;ftag=uloc-559adce7-19bb-91-ac5e6b73-0305b6a6;lr=on><br>
> ><br>
> > Via: SIP/2.0/WSS<br>
> 54.189.6.185:5061;branch=z9hG4bK68e6.7c6248942a593cf21da989d27e9b4cf3.0<br>
> ><br>
> > Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0<br>
> ><br>
> > From:<br>
> sip:pinger@sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-0305b6a6<br>
> ><br>
> > To: sip:d62a2g56@ug5tmpr4sfhc.invalid;transport=ws<br>
> ><br>
> > Call-ID: <a href="mailto:8c334a51-467f0854-ddf23e2@127.0.0.1">8c334a51-467f0854-ddf23e2@127.0.0.1</a><br>
> <mailto:<a href="mailto:8c334a51-467f0854-ddf23e2@127.0.0.1">8c334a51-467f0854-ddf23e2@127.0.0.1</a>><br>
> ><br>
> > CSeq: 1 OPTIONS<br>
> ><br>
> > Content-Length: 0<br>
> ><br>
> ><br>
> ><br>
> ><br>
> > " " +25s" jssip.js:21621<br>
> > "JsSIP:Transport " "sending WebSocket message:<br>
> ><br>
> > SIP/2.0 200 OK<br>
> ><br>
> > Via: SIP/2.0/WSS<br>
> 54.189.6.185:5061;branch=z9hG4bK68e6.7c6248942a593cf21da989d27e9b4cf3.0<br>
> ><br>
> > Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0<br>
> ><br>
> > To: sip:d62a2g56@ug5tmpr4sfhc.invalid;transport=ws;tag=0vv3pidftj<br>
> ><br>
> > From:<br>
> sip:pinger@sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-0305b6a6<br>
> ><br>
> > Call-ID: <a href="mailto:8c334a51-467f0854-ddf23e2@127.0.0.1">8c334a51-467f0854-ddf23e2@127.0.0.1</a><br>
> <mailto:<a href="mailto:8c334a51-467f0854-ddf23e2@127.0.0.1">8c334a51-467f0854-ddf23e2@127.0.0.1</a>><br>
> ><br>
> > CSeq: 1 OPTIONS<br>
> ><br>
> > Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS<br>
> ><br>
> > Accept: application/sdp, application/dtmf-relay<br>
> ><br>
> > Supported: outbound<br>
> ><br>
> > Content-Length: 0<br>
> ><br>
> ><br>
> ><br>
> ><br>
><br>
><br>
><br>
</blockquote></div><br></div>