[Spce-user] Web RTC error

[Brian Quandt]

Daniel,

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.

Restoring everything back to it's original I'm back again, with users able
to connect to one another, and the failured in webrtc...

But as I said you have me hopefully looking in the right direction.

Yours truly,
Brian


On Wed, Jul 8, 2015 at 1:47 PM, Daniel Grotti <dgrotti at sipwise.com> wrote:

> 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
> >
> >
> >
> >
>
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