[Spce-user] Web RTC error
Daniel Grotti
dgrotti at sipwise.com
Fri Jul 10 08:16:23 EDT 2015
Hi,
it's hard to tell that.
You should analyze the trace on your web browser with the network tools
provided by chrome.
Usually problems is related to some video codec set by the callee (and
not by the caller) or you may change the parameter in spce for the
"transport_protocol".
--
Daniel Grotti
VoIP Engineer
Sipwise GmbH
Europaring F15 | 2345 Brunn am Gebirge, Austria | www.sipwise.com
On 07/09/2015 09:22 PM, Brian Quandt wrote:
> Just tried jssip to jssip via two users configured as Daniel suggested.
>
> I get a WebRTC error on one chrome browser (the callee), and a
> "incompatible sdp" on the caller.
>
> Again, both chrome browsers, same version as Daniel is using.
>
> Yours truly,
> Brian
>
>
> On Thu, Jul 9, 2015 at 11:53 AM, Brian Quandt <brian.quandt at gmail.com
> <mailto:brian.quandt at gmail.com>> wrote:
>
> Paritally figured this one out.
>
> "Denied Media", was because jssip had "video" checked by default.
> Chrome can't deal with my camera (because my camera happens to be
> some new dev stuff I'm working on). So user issue on that one.
> Sorry for the question
>
> I'm now able to connect between jssip via my own sip/ws server per
> your notes thanks you.
>
> Howeveer, I dont' "hear" anything.
>
> Obviously could still be driver related in chrome to my audio
> device... checking that now. But in testing this on OSX/Yosemite,
> ie one using jssip and the other using jitsi, still the same
> problem, ie they connect but no audio (jssip does prompt in each
> case for microphone access which of course I say accept).
>
> Yours truly,
> Brian
>
>
> On Thu, Jul 9, 2015 at 11:37 AM, Brian Quandt
> <brian.quandt at gmail.com <mailto:brian.quandt at gmail.com>> wrote:
>
> Thanks.
>
> I almost have things working (same chrome as you are using).
>
> I'm getting a chrome issue, OS Ubuntu 14.04. "Denied Media
> Access".
>
> If I'm using chrome under OSX (tryit.jssip.net
> <http://tryit.jssip.net>) to someone jitsi, seems to have worked
> briefly (just not tested throughly yet).
>
> Dont' think this has anything to do with spce, but would
> appreciate anyones thoughts of chrome under ubuntu (googling
> seems to show this issue as well on other non spce things, but
> don't know the work around).
>
> Yours truly,
> Brian
>
>
> On Thu, Jul 9, 2015 at 12:01 AM, Daniel Grotti
> <dgrotti at sipwise.com <mailto:dgrotti at sipwise.com>> wrote:
>
> 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 <http://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>
> <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>
> <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>
> <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>
> > <mailto: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>
> > <mailto: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>
> > <mailto: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>
> <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> <http://0.0.0.0/0>
> > > HTTPS
> > > TCP
> > > 443
> > > 0.0.0.0/0 <http://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> <http://0.0.0.0/0>
> > > Custom TCP Rule
> > > TCP
> > > 1443
> > > 0.0.0.0/0 <http://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> <http://0.0.0.0/0>
> > > Custom TCP Rule
> > > TCP
> > > 5060
> > > 0.0.0.0/0 <http://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> <http://0.0.0.0/0>
> > > Custom UDP Rule
> > > UDP
> > > 5060
> > > 0.0.0.0/0 <http://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> <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>
> > <http://tryit.jssip.net>, with folowing details:
> > > name: quandt
> > > sip uri: sip:quandt at sip.autodcp.com
> <mailto:sip%3Aquandt at sip.autodcp.com>
> > <mailto:sip%3Aquandt at sip.autodcp.com
> <mailto:sip%253Aquandt at sip.autodcp.com>>
> > > password: ******
> > > ws uri: wss://sip.autodcp.com:1443/wss/sip/
> <http://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>
> > <mailto: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>
> > <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>
> > <mailto:sip%3Ad62a2g56 at sip.autodcp.com
> <mailto:sip%253Ad62a2g56 at sip.autodcp.com>>>;tag=2mftbnm4nm
> > >
> > > From: <sip:0991002 at sip.autodcp.com
> <mailto:sip%3A0991002 at sip.autodcp.com>
> > <mailto:sip%3A0991002 at sip.autodcp.com
> <mailto:sip%253A0991002 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>
> > <mailto:sip%3A0991002 at sip.autodcp.com
> <mailto:sip%253A0991002 at sip.autodcp.com>>>;tag=5A6A45EC-559D727B0008760D-D4D52700
> > >
> > > To: <sip:d62a2g56 at sip.autodcp.com
> <mailto:sip%3Ad62a2g56 at sip.autodcp.com>
> > <mailto:sip%3Ad62a2g56 at sip.autodcp.com
> <mailto:sip%253Ad62a2g56 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>
> > <mailto: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>
> > <mailto: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>
> > <mailto: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>
> > <mailto: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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