[Spce-user] Web RTC error

Wed Jul 8 16:47:03 EDT 2015

Hi Brian,
Maybe this could be a good start: https://www.linkedin.com/pulse/how-enable-webrtc-sipprovider-daniel-grotti?_mSplash=1

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.

Daniel

On Jul 8, 2015 9:19 PM, Brian Quandt <brian.quandt at gmail.com> wrote:
>
> Trying to get things working and am stumbling.  Maybe someone can help me a bit?
>
> Right now, I just want to get things working, ie do a simple test using jssip.net, based on the AWS AMI image built by sipwise, ie sip:provider CE AMI mr3.8.2, image id:  ami-17142e27 (us west 2)
>
> Here's my steps so far:
>
> 1) got the ec2 instance running
> 2) configured the ec2 security group/ports as below:
>
> HTTP
> TCP
> 80
> 0.0.0.0/0
> HTTPS
> TCP
> 443
> 0.0.0.0/0
> Custom TCP Rule
> TCP
> 1080
> 0.0.0.0/0
> Custom TCP Rule
> TCP
> 1443
> 0.0.0.0/0
> Custom TCP Rule
> TCP
> 2443
> 0.0.0.0/0
> Custom TCP Rule
> TCP
> 5060
> 0.0.0.0/0
> Custom TCP Rule
> TCP
> 5061
> 0.0.0.0/0
> Custom UDP Rule
> UDP
> 5060
> 0.0.0.0/0
> Custom UDP Rule
> UDP
> 5061
> 0.0.0.0/0
> ssh is configured for my machine only (obviously)
>
> 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).
>
> 4) launched firefox under linux going to tryit.jssip.net, with folowing details:
> name:  quandt
> sip uri:  sip:quandt at sip.autodcp.com
> password:  ******
> ws uri:  wss://sip.autodcp.com:1443/wss/sip/
>
> Which got me to the jssip demo page both connected and registered just fine.
>
> 5) on a mac launched zoiper and logged into another account on my sip server
>
> 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.
>
> Any thoughts?
>
> Yours truly,
> Brian
>
>
>
> " " +2s" jssip.js:21621
> "JsSIP:Transport " "sending WebSocket message:
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/WSS 54.189.6.185:5061;branch=z9hG4bKdf8e.4e2788dc9d9e3774bc01623f84011d82.0
>
> Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0
>
> To: sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws;tag=je09o4s8o3
>
> From: sip:pinger at sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-8205b6a6
>
> Call-ID: 8c334a51-c57f0854-1af23e2 at 127.0.0.1
>
> CSeq: 1 OPTIONS
>
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
>
> Accept: application/sdp, application/dtmf-relay
>
> Supported: outbound
>
> Content-Length: 0
>
>
>
>
> " " +16ms" jssip.js:21621
> "JsSIP:NonInviteServerTransaction " "Timer J expired for transaction z9hG4bKdf8e.4e2788dc9d9e3774bc01623f84011d82.0" " +3ms" jssip.js:21621
> "JsSIP:RTCSession " "answer()" " +1s" jssip.js:21621
> "JsSIP:Dialog " "dialog MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-12mftbnm4nm5A6A45EC-559D727B0008760D-D4D52700  changed to CONFIRMED state" " +0ms" jssip.js:21621
> "rtcninja:RTCPeerConnection " "new | pcConfig: " Object { iceServers: Array[1], gatheringTimeout: 2000 } " +3ms" jssip.js:21621
> "rtcninja:RTCPeerConnection " "setConfigurationAndOptions | processed pcConfig: " Object { iceServers: Array[1] } " +1ms" jssip.js:21621
> "rtcninja:Adapter " "getUserMedia() | constraints: " Object { audio: true, video: false } " +93ms" jssip.js:21621
> Invalid URI. Load of media resource  failed. tryit.jssip.net
> "rtcninja:Adapter " "getUserMedia() | success" " +2s" jssip.js:21621
> "rtcninja:RTCPeerConnection " "addStream() | stream: [object LocalMediaStream]" " +0ms" jssip.js:21621
> "rtcninja:RTCPeerConnection " "setRemoteDescription()" " +1ms" jssip.js:21621
> "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
> "JsSIP:Transport " "sending WebSocket message:
>
> SIP/2.0 488 Not Acceptable Here
>
> Via: SIP/2.0/WSS 54.189.6.185:5061;branch=z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0
>
> Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKuN~pJa9J;rport=5080
>
> To: <sip:d62a2g56 at sip.autodcp.com>;tag=2mftbnm4nm
>
> From: <sip:0991002 at sip.autodcp.com>;tag=5A6A45EC-559D727B0008760D-D4D52700
>
> Call-ID: MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-1
>
> CSeq: 10 INVITE
>
> Supported: timer,ice,outbound
>
> Content-Length: 0
>
>
>
>
> " " +0ms" jssip.js:21621
> "JsSIP:RTCSession " "session failed" " +1ms" jssip.js:21621
> "JsSIP:RTCSession " "close()" " +0ms" jssip.js:21621
> "rtcninja:RTCPeerConnection " "close()" " +0ms" jssip.js:21621
> "rtcninja:RTCPeerConnection " "oniceconnectionstatechange() | iceConnectionState: closed" " +0ms" jssip.js:21621
> "JsSIP:RTCSession " "close() | closing local MediaStream" " +0ms" jssip.js:21621
> "rtcninja:Adapter " "closeMediaStream() | calling stop() on all the MediaStreamTrack" " +1ms" jssip.js:21621
> "JsSIP:Dialog " "dialog MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-12mftbnm4nm5A6A45EC-559D727B0008760D-D4D52700 deleted" " +4ms" jssip.js:21621
> "rtcninja:RTCPeerConnection " "onsignalingstatechange() | signalingState: closed" " +5ms" jssip.js:21621
> "JsSIP:Transport " "received WebSocket text message:
>
> ACK sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws SIP/2.0
>
> Max-Forwards: 70
>
> 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>
>
> 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>
>
> Via: SIP/2.0/WSS 54.189.6.185:5061;branch=z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0
>
> Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKuN~pJa9J;rport=5080
>
> From: <sip:0991002 at sip.autodcp.com>;tag=5A6A45EC-559D727B0008760D-D4D52700
>
> To: <sip:d62a2g56 at sip.autodcp.com>;tag=2mftbnm4nm
>
> Call-ID: MDQ4M2M5YzQ1ODhmNDM5ZTZhNTIwMTNjZWQ2Y2I5YTU._b2b-1
>
> CSeq: 10 ACK
>
> Content-Length: 0
>
> Route: <sip:10.220.196.230:32769;transport=ws>
>
>
>
>
> " " +31ms" jssip.js:21621
> "JsSIP:Transport " "received WebSocket text message:
>
> OPTIONS sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws SIP/2.0
>
> Max-Forwards: 70
>
> 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>
>
> 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>
>
> Via: SIP/2.0/WSS 54.189.6.185:5061;branch=z9hG4bK8d0f.6a6365b82f7b9a11de527fd7f0d652bd.0
>
> Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0
>
> From: sip:pinger at sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-c205b6a6
>
> To: sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws
>
> Call-ID: 8c334a51-067f0854-fbf23e2 at 127.0.0.1
>
> CSeq: 1 OPTIONS
>
> Content-Length: 0
>
>
>
>
> " " +27s" jssip.js:21621
> "JsSIP:Transport " "sending WebSocket message:
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/WSS 54.189.6.185:5061;branch=z9hG4bK8d0f.6a6365b82f7b9a11de527fd7f0d652bd.0
>
> Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0
>
> To: sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws;tag=5r8pi0ggps
>
> From: sip:pinger at sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-c205b6a6
>
> Call-ID: 8c334a51-067f0854-fbf23e2 at 127.0.0.1
>
> CSeq: 1 OPTIONS
>
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
>
> Accept: application/sdp, application/dtmf-relay
>
> Supported: outbound
>
> Content-Length: 0
>
>
>
>
> " " +7ms" jssip.js:21621
> "JsSIP:NonInviteServerTransaction " "Timer J expired for transaction z9hG4bK8d0f.6a6365b82f7b9a11de527fd7f0d652bd.0" " +0ms" jssip.js:21621
> "JsSIP:InviteServerTransaction " "Timer H expired for transaction z9hG4bKd4f3.10c8fa1639929c67088fb474ef232c46.0" " +5s" jssip.js:21621
> "JsSIP:Transport " "received WebSocket text message:
>
> OPTIONS sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws SIP/2.0
>
> Max-Forwards: 70
>
> 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>
>
> 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>
>
> Via: SIP/2.0/WSS 54.189.6.185:5061;branch=z9hG4bK68e6.7c6248942a593cf21da989d27e9b4cf3.0
>
> Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0
>
> From: sip:pinger at sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-0305b6a6
>
> To: sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws
>
> Call-ID: 8c334a51-467f0854-ddf23e2 at 127.0.0.1
>
> CSeq: 1 OPTIONS
>
> Content-Length: 0
>
>
>
>
> " " +25s" jssip.js:21621
> "JsSIP:Transport " "sending WebSocket message:
>
> SIP/2.0 200 OK
>
> Via: SIP/2.0/WSS 54.189.6.185:5061;branch=z9hG4bK68e6.7c6248942a593cf21da989d27e9b4cf3.0
>
> Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=0
>
> To: sip:d62a2g56 at ug5tmpr4sfhc.invalid;transport=ws;tag=0vv3pidftj
>
> From: sip:pinger at sipwise.local;tag=uloc-559adce7-19bb-91-ac5e6b73-0305b6a6
>
> Call-ID: 8c334a51-467f0854-ddf23e2 at 127.0.0.1
>
> CSeq: 1 OPTIONS
>
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
>
> Accept: application/sdp, application/dtmf-relay
>
> Supported: outbound
>
> Content-Length: 0
>
>
>
>