[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
>         >
>         >
>         >
>         >
> 
> 
> 



More information about the Spce-user mailing list