[SR-Users] Kamailio + FreeSwitch + WebRTC

[Yuriy Gorlichenko]

https://github.com/2600hz/kazoo-configs-kamailio/blob/master/kamailio/websockets-role.cfg

2018-06-21 18:35 GMT+03:00 Emanuel Gianico <emanuelgianico at gmail.com>:

> I don't think so... The idea is to use FS as a media server. Why I would
> use RTPEngine? FS offers the same and more:
>
> - WebRTC support
> - VoiceMail Server
> - Queues
> - IVR
> - Announcements
>
> And so on...
>
> The idea is to use Kamailio in front and throw all media related stuff to
> FS.
>
> There is a way to accomplish this? I searched a lot but I couldn't find
> nothing about it, only about Kamailio and RTPProxy (or RTPEngine)
>
> Best regards,
> Emanuel.
>
> El vie., 15 de jun. de 2018 09:32, Pan Christensen <
> pan.christensen at phonect.no> escribió:
>
>> Or maybe FreeSwitch is redundant if you use rtpengine…
>>
>>
>>
>> With kind regards
>> *Pan B. Christensen*
>> Developer
>> Phonect AS
>>
>>
>>
>> *From:* sr-users <sr-users-bounces at lists.kamailio.org> * On Behalf Of *Emanuel
>> Gianico
>> *Sent:* fredag 15. juni 2018 13:29
>> *To:* Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
>> *Subject:* Re: [SR-Users] Kamailio + FreeSwitch + WebRTC
>>
>>
>>
>> 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/kamai
>> lio/kamailio.cfg
>> but I blamed my failure on an old rtpengine :)
>>
>>
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>>
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>>
>
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