18 Jul
2023
18 Jul
'23
11:42 a.m.
I wouldn't do a delete in that situation; it might be simpler to just do another
rtpengine_offer() on a <Call-ID, From-tag> tuple that RTPEngine already knows about,
just as one would in a reinvite. I believe rtpengine_manage() would cover this situation
as well.
And yes, then just t_relay() anew!
On Jul 18, 2023, at 11:41 AM, Barry Flanagan
<barry(a)flantel.com> wrote:
On 18/07/2023 16:33, Alex Balashov wrote:
--
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800
Barry,
Might it not be simpler to follow Richard's first suggestion in that thread, and
simply catch the 488 response in a failure_route and re-issue a non-SRTP offer on another
branch?
hi Alex,
Might be. I am just unsure about how to instruct rtpengine to re-send the plain RTP
invite. Do I do a rtpengine_delete and then restart with rtpengine_manage() then relay
again?
Thanks.
-Barry
-- Alex
> On Jul 18, 2023, at 11:05 AM, Barry Flanagan <barry(a)flantel.com> wrote:
>
> Hi,
>
> I have implemented a WebRTC proxy using Kamailio and rtpengine, in front
> of Asterisk servers. I have an issue where I do not know in advance
> whether the Asterisk client is set to use RTP or SRTP.
>
> My implementation expects the web side to be SRTP and the Asterisk side
> plain RTP. However, if the asterisk side requires SRTP I receive a 488
> Not Acceptable here. The same is true when an SRTP client from the
> Asterisk side dials out to the webrtc client, I need to catch whether
> the request is SRTP or RTP and generate rtpengine offer accordingly.
>
> This was discussed in github, with a description of a solution here:
> https://github.com/sipwise/rtpengine/issues/435#issuecomment-391937398 -
> however I am unsure how to implement this.
>
> Does anyone have an example of how I can change the rtpengine SDP in
> such a case?
>
> Thanks
> --
> -Barry
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--
-Barry