<div dir="ltr"><div><div><div><div><div>Daniel,<br><br></div>Thanks for the reply. It got me looking, but the instructions as references actually was a step backwards. Lost all connectivity for all users, plus the mods to the user I'd set for webrtc couldn't even connect/register per jitsi.<br><br></div>Restoring everything back to it's original I'm back again, with users able to connect to one another, and the failured in webrtc... <br><br></div>But as I said you have me hopefully looking in the right direction.<br><br></div>Yours truly,<br></div>Brian<br><br></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Jul 8, 2015 at 1:47 PM, 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>
Maybe this could be a good start: <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 transport_protocol in ngcp toeard s the webrtc client in a 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>> wrote:<br>
><br>
> Trying to get things working and am stumbling. Maybe someone can help me a bit?<br>
><br>
> Right now, I just want to get things working, ie do a simple test using <a href="http://jssip.net" rel="noreferrer" target="_blank">jssip.net</a>, based on the AWS AMI image built by sipwise, ie sip:provider CE AMI mr3.8.2, image 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><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><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><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><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><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><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><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><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><br>
> ssh is configured for my machine only (obviously)<br>
><br>
> 3) got a proper ssl cert from godaddy, change all my sslcerfile and sslkey files in config.yml appropriately, and made sure kamailio tls is enabled (which it is by default in the 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>, with folowing details:<br>
> name: quandt<br>
> sip uri: <a href="mailto:sip%3Aquandt@sip.autodcp.com">sip:quandt@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>
><br>
> Which got me to the jssip demo page both connected and registered just fine.<br>
><br>
> 5) on a mac launched zoiper and logged into another account on my sip server<br>
><br>
> 6) tried to call from one to the other. Got a ring from one ot the other to work, on the jssip demo page, when I ansewred, I get promoted to share my microphone, which I acknowlege, and them get a WebRTC error right away. Below is part of the 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 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: 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>
><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 transaction z9hG4bKdf8e.4e2788dc9d9e3774bc01623f84011d82.0" " +3ms" jssip.js:21621<br>
> "JsSIP:RTCSession " "answer()" " +1s" jssip.js:21621<br>
> "JsSIP:Dialog " "dialog MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-12mftbnm4nm5A6A45EC-559D727B0008760D-D4D52700 changed to CONFIRMED state" " +0ms" jssip.js:21621<br>
> "rtcninja:RTCPeerConnection " "new | pcConfig: " Object { iceServers: Array[1], gatheringTimeout: 2000 } " +3ms" jssip.js:21621<br>
> "rtcninja:RTCPeerConnection " "setConfigurationAndOptions | processed pcConfig: " Object { iceServers: Array[1] } " +1ms" jssip.js:21621<br>
> "rtcninja:Adapter " "getUserMedia() | constraints: " Object { 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>
> "rtcninja:Adapter " "getUserMedia() | success" " +2s" jssip.js:21621<br>
> "rtcninja:RTCPeerConnection " "addStream() | stream: [object LocalMediaStream]" " +0ms" jssip.js:21621<br>
> "rtcninja:RTCPeerConnection " "setRemoteDescription()" " +1ms" jssip.js:21621<br>
> "rtcninja:ERROR:RTCPeerConnection " "setRemoteDescription() | error:" " +1ms" Object { name: "INVALID_SESSION_DESCRIPTION", message: "Could not negotiate media lines; cause = NO_DTLS_FINGERPRINT | SDP Parsing Error: Warning: No network type specified in comediadir attribute.", __exposedProps__: 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 54.189.6.185:5061;branch=z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0<br>
><br>
> Via: SIP/2.0/UDP 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>>;tag=2mftbnm4nm<br>
><br>
> From: <<a href="mailto:sip%3A0991002@sip.autodcp.com">sip:0991002@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() | iceConnectionState: closed" " +0ms" jssip.js:21621<br>
> "JsSIP:RTCSession " "close() | closing local MediaStream" " +0ms" jssip.js:21621<br>
> "rtcninja:Adapter " "closeMediaStream() | calling stop() on all the MediaStreamTrack" " +1ms" jssip.js:21621<br>
> "JsSIP:Dialog " "dialog MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-12mftbnm4nm5A6A45EC-559D727B0008760D-D4D52700 deleted" " +4ms" jssip.js:21621<br>
> "rtcninja:RTCPeerConnection " "onsignalingstatechange() | 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: <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: <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 54.189.6.185:5061;branch=z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0<br>
><br>
> Via: SIP/2.0/UDP 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>>;tag=5A6A45EC-559D727B0008760D-D4D52700<br>
><br>
> To: <<a href="mailto:sip%3Ad62a2g56@sip.autodcp.com">sip:d62a2g56@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: <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: <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 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: 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>
><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 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: 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>
><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 transaction z9hG4bK8d0f.6a6365b82f7b9a11de527fd7f0d652bd.0" " +0ms" jssip.js:21621<br>
> "JsSIP:InviteServerTransaction " "Timer H expired for transaction z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0" " +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: <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: <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 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: 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>
><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 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: 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>
><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>
</blockquote></div><br></div>