[SR-Users] Kamailio + FreeSwitch + WebRTC

[Emanuel Gianico]

I'm going to investigate Kazoo samples as Gorlichenko suggested because I
think using RTPEngine or rtp proxy seems redundant/unnecesary to me since
FreeSwitch fully supports WebRTC

El jue., 14 de jun. de 2018 17:42, Yuriy Gorlichenko <ovoshlook at gmail.com>
escribió:

> You can watch at the kazoo project examples if you want to avoid rtp proxy
>
> On Thu, Jun 14, 2018, 23:26 Daniel Tryba <d.tryba at pocos.nl> wrote:
>
>> On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
>> > From the logs I see the jssip throw this error:
>> >
>> > "Failed to set remote offer sdp: Called with SDP without DTLS
>> fingerprint."
>> >
>> > I would like to avoid RTPEngine, because from what I understand,
>> FreeSwitch
>> > can handle the media part.
>>
>> IIRC I got the same error in my tries to transcode/bridge SIP over TLS
>> with SRTP to just plain old SIP with RTP. I haven't put any effort in it
>> to get it working. You'll need to play around with rtpengine
>> offer/answer, I based my test on
>>
>> https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg
>> but I blamed my failure on an old rtpengine :)
>>
>>
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