[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

[John Nash]

OK here is the log
https://gist.github.com/johnnash13/0d2cb5238f3551cd3a8c6b4e638dd744

Sorry took me a while to convert wireshark trace to text file.

My freeswitch is running on private IP (127.0.0.1) and opensips I run on
both public and private so that for outside world opensips is the only
public IP they see. In proxy log I pasted Opensips ===> Freeswitch logs and
back.






On Fri, Jun 24, 2016 at 12:43 AM, Eric Tamme <eric at uphreak.com> wrote:

> No - it's annoying to look at a trace that's had information removed and
> try and piece together whats happening.  Your paranoid side is wrong, sorry.
>
> -Eric
>
>
> On 06/23/2016 01:06 PM, Patrick Wakano wrote:
>
> 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>
>> 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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>>
>>
>>
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>>
>>
>
>
> _______________________________________________
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>
>
>
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