[Spce-user] Web RTC error
Brian Quandt
brian.quandt at gmail.com
Wed Jul 8 19:30:01 EDT 2015
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 , 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 (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 , 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> 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> 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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