[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[John Nash]

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> wrote:

> Hey John,
>
> Please paste a full UNALTERED sip trace into a gist (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> 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>
>> 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?
>>
>>
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>>
>>
>>
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>>
>
>
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