[Spce-user] Web RTC error
Brian Quandt
brian.quandt at gmail.com
Thu Jul 9 15:22:50 EDT 2015
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>
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>
> 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) 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>
>> 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
>>>
>>> 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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