[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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