[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[Eric Tamme]

I mean you can use a private gist, but you will be publishing the link 
in a public email list.  In general I personally dont believe revealing 
ip addresses etc. is any problem - to put my money where my mouth is 
here is a gist link to an unaltered SIP trace on my server :)

https://gist.github.com/etamme/b864010448a29007b7e0457682e81d52

-Eric

On 06/23/2016 12:23 PM, John Nash wrote:
> Ok i am ready with logs. About gist may I use private option as traces 
> have our IPs, user
>
> On Thu, Jun 23, 2016 at 10:32 PM, Eric Tamme <eric at uphreak.com 
> <mailto:eric at uphreak.com>> wrote:
>
>     Hey John,
>
>     Please paste a full UNALTERED sip trace into a gist
>     (gist.github.com <http://gist.github.com>) from the proxy servers
>     perspective and provide a link so that we can see what comes in,
>     and what goes out from both sides.
>
>     EG: ngrep -qtd any -W byline port 5060
>
>     This will show us the traffic that is leaving the proxy destined
>     for the Freeswitch box, and what the freeswitch box sends back.
>
>     Also - you can look in your browsers console log and provide the
>     SIP trace from there in a seperate gist, so that we can see what
>     opensips sends back up to your browser.
>
>     -Eric
>
>
>>     Am I using correct sip.js example? I copied it to my server and
>>     accessing it using https: (used letsencrypt)
>>
>>     On Thu, Jun 23, 2016 at 7:58 PM, Eric Tamme <eric at uphreak.com
>>     <mailto:eric at uphreak.com>> wrote:
>>
>>         1. I would suggest using SIP.js -
>>         https://github.com/onsip/SIP.js it is a much more active
>>         project that sipml5.
>>
>>         2. Im guessing that you are not properly passing flags to
>>         RTPEngine.  If you want to have DTLS-SRTP between the
>>         browser, and plain RTP/AVP between RTPEngine and freeswitch,
>>         you need to "offer" rtp/avp to freeswitch, and "answer"
>>         dtls-srtp back up to the browser.
>>
>>         the offer to freeswitch would be:
>>
>>                  $var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
>>
>>         and the answer back up to the browswer would be:
>>
>>                  $var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
>>
>>
>>         -Eric
>>
>>
>>
>>         On 06/23/2016 08:20 AM, John Nash wrote:
>>>         I am following
>>>         http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 and
>>>         trying to test a call
>>>
>>>         sipml5 ----------->Opensips + rtpengine --------> SIP end
>>>         point (Freeswitch)
>>>
>>>         But I do not have any audio on both sides. I see this error
>>>         at rtpengine log "SRTP output wanted, but no crypto suite
>>>         was negotiated"
>>>
>>>         Anyone tested this scenario positive?
>>>
>>>
>>>         _______________________________________________
>>>         Users mailing list
>>>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>         _______________________________________________
>>         Users mailing list
>>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>>
>>     _______________________________________________
>>     Users mailing list
>>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20160623/ba14fe53/attachment.htm>