[OpenSIPS-Users] Opensips + rtpengine + Sipml5 webrtc

Eric Tamme eric at uphreak.com
Thu Jun 23 21:54:49 CEST 2016


And are you forcing RTPengine to act as an ice light client?  It looks 
like you are gettin a single ICE candidate in the answer back from 
freeswitch which would indicate that you are.

I'd check your chrome webrtc statistics to see if tis failed to do do 
ice/stun negotiation on the 183.  In general the signalling looks good.

I think you may have an error on your Freeswitch side - some thing that 
is trying to force it to use SRTP all the time, even though the 
signalling has requested plain RTP (to freeswitch).

I think you should ask in #freeswitch on freenode at this point.

-Eric

On 06/23/2016 01:42 PM, John Nash wrote:
> Actually the issue is i hear no audio on either side and just after 
> session progress (I guess when media starts coming from remote media 
> server) i see error  "SRTP output wanted, but no crypto suite was 
> negotiated"
>
> I had also checked media logs i could see RTP packets being sent from 
> freeswitch to RTPengine IP but there was no packet at all just after 
> that. Ideally after RTP packet from freeswitch to rtpengine, Rtpengine 
> should send that packet to browser using wss?
>
> On Fri, Jun 24, 2016 at 1:05 AM, Eric Tamme <eric at uphreak.com 
> <mailto:eric at uphreak.com>> wrote:
>
>     So - i dont see a problem here - Chrome is getting
>     UDP/TLS/RTP/SAVPF and Freeswitch is getting RTP/AVP.  Freeswitch
>     responded to the offer in the invite with an answer in the 183,
>     and in the 200.  What is the failure you are seeing, and where is
>     it happening (in freeswitch? in the browser?)
>
>     The only thing that looks bad is that you are retransmitting the
>     ACK which FS either ... doesnt like, or is never getting,  because
>     it keeps retransmitting the 200, which is why you get a 481 when
>     you send BYE.
>
>     -Eric
>
>
>     On 06/23/2016 01:24 PM, John Nash wrote:
>>     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
>>     <mailto: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 <mailto: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 <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?
>>>>>>
>>>>>>
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>>>>>
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>>>>>
>>>>>
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