[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[Patrick Wakano]

my paranoic side would recommend to hide/change private informations,
specially any authentication line that might appear... this is certainly a
sort of social engineering threat we should worry...
better be safe than sorry....


On Thu, Jun 23, 2016 at 3:31 PM, Eric Tamme <eric at uphreak.com> wrote:

> 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> 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>
>> eric at uphreak.com> wrote:
>>
>>> 1. I would suggest using SIP.js - <https://github.com/onsip/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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>>
>>
>>
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>
>
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