[Spce-user] Web RTC error
Daniel Grotti
dgrotti at sipwise.com
Thu Jul 9 03:01:44 EDT 2015
Hi Brian,
this is a working configuration for Chrome Version 43.0.2357.65:
ws://yourip:5060/ws, wss://yourip:5061/ws, wss://yourip:1443/wss/sip/
* WebRTC Subscribers -> Details -> Preferences -> NAT and Media Flow Control
- 'use_rtpproxy:' Always with rtpproxy as additional/only ICE candidate
- 'transport_protocol:' RTP/SAVPF (encrypted SRTP with RTCP feedback)
(for Chrome Version 43.0.2357.65)
* Domain -> Details -> Preferences -> NAT and Media Flow Control
- 'transport_protocol:' RTP/AVP (Plain RTP)
--
Daniel Grotti
VoIP Engineer
Sipwise GmbH
Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com
On 07/09/2015 01:30 AM, Brian Quandt wrote:
> What does this mean in the rtp.log?
>
> I've not yet been able to get jitsi or jssip to work completely. But I
> can get users to login/register on each clietn, and they can call each
> other (but upon asnwering, nothing, or a webrtc failure in jssip).
>
> Maybe the "unknown codec" is the issue?
>
> Yours truly,
> Brian
>
>
> A89F8-D5A5F700'
> Jul 8 23:23:45 ip-10-220-196-230 rtpengine[6430]:
> [a9e2e70e7cb3bb435b081bc5e2469ee2 at 0:0:0:0:0:0:0:0] ------ Media #1
> (audio over UDP/TLS/RTP/SAVPF) using unknown codec
> Jul 8 23:23:45 ip-10-220-196-230 rtpengine[6430]:
> [a9e2e70e7cb3bb435b081bc5e2469ee2 at 0:0:0:0:0:0:0:0] --------- Port 32832
> <> 157.254.210.17:5000 <http://157.254.210.17:5000> , 0 p, 0 b, 0 e,
> 1436397570 last_packet
> Jul 8 23:23:45 ip-10-220-196-230 rtpengine[6430]:
> [a9e2e70e7cb3bb435b081bc5e2469ee2 at 0:0:0:0:0:0:0:0] --------- Port 32833
> <> 157.254.210.17:5001 <http://157.254.210.17:5001> (RTCP), 0 p, 0 b,
> 0 e, 1436397570 last_packet
> Jul 8 23:23:45 ip-10-220-196-230 rtpengine[6430]:
> [a9e2e70e7cb3bb435b081bc5e2469ee2 at 0:0:0:0:0:0:0:0] ------ Media #2
> (video over UDP/TLS/RTP/SAVPF) using unknown codec
> Jul 8 23:23:45 ip-10-220-196-230 rtpengine[6430]:
> [a9e2e70e7cb3bb435b081bc5e2469ee2 at 0:0:0:0:0:0:0:0] --------- Port 32846
> <> 157.254.210.17:5016 <http://157.254.210.17:5016> , 0 p, 0 b, 0 e,
> 1436397570 last_packet
>
>
> On Wed, Jul 8, 2015 at 3:03 PM, Brian Quandt <brian.quandt at gmail.com
> <mailto:brian.quandt at gmail.com>> wrote:
>
> 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
> <mailto: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
> <mailto: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 <http://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 <http://0.0.0.0/0>
> > HTTPS
> > TCP
> > 443
> > 0.0.0.0/0 <http://0.0.0.0/0>
> > Custom TCP Rule
> > TCP
> > 1080
> > 0.0.0.0/0 <http://0.0.0.0/0>
> > Custom TCP Rule
> > TCP
> > 1443
> > 0.0.0.0/0 <http://0.0.0.0/0>
> > Custom TCP Rule
> > TCP
> > 2443
> > 0.0.0.0/0 <http://0.0.0.0/0>
> > Custom TCP Rule
> > TCP
> > 5060
> > 0.0.0.0/0 <http://0.0.0.0/0>
> > Custom TCP Rule
> > TCP
> > 5061
> > 0.0.0.0/0 <http://0.0.0.0/0>
> > Custom UDP Rule
> > UDP
> > 5060
> > 0.0.0.0/0 <http://0.0.0.0/0>
> > Custom UDP Rule
> > UDP
> > 5061
> > 0.0.0.0/0 <http://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
> <http://tryit.jssip.net>, with folowing details:
> > name: quandt
> > sip uri: sip:quandt at sip.autodcp.com
> <mailto:sip%3Aquandt at sip.autodcp.com>
> > password: ******
> > ws uri: wss://sip.autodcp.com:1443/wss/sip/
> <http://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
> <mailto: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
> <http://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
> <mailto:sip%3Ad62a2g56 at sip.autodcp.com>>;tag=2mftbnm4nm
> >
> > From: <sip:0991002 at sip.autodcp.com
> <mailto:sip%3A0991002 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
> <mailto:sip%3A0991002 at sip.autodcp.com>>;tag=5A6A45EC-559D727B0008760D-D4D52700
> >
> > To: <sip:d62a2g56 at sip.autodcp.com
> <mailto:sip%3Ad62a2g56 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
> <mailto: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
> <mailto: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
> <mailto: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
> <mailto: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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