20 May
2016
20 May
'16
9:49 a.m.
Yes, now it is working both ways. I was missing the rtpengine and right configuration.
I will do some further testing to see if everything is functional.
Thanks,
To: sr-users(a)lists.sip-router.org
From: rfuchs(a)sipwise.com
Date: Thu, 19 May 2016 08:22:26 -0400
Subject: Re: [SR-Users] Browser WebRTC transcoder
On 05/19/2016 04:52 AM, Moacir Ferreira wrote:
...
So the Grandstream offers a lot of codecs but
will get a "Not Found"
from Kamailio. Look in the other way:
That would be a SIP signalling (e.g. Kamailio config) problem. Perhaps a
missing registration.
Here the Grandstream says "Media type not
available". As I am not a real
SIP guy, I got no clue why does not work!
This you can solve with rtpengine. The required codecs (PCM) are there,
you just need to break the encryption (RTP <> SRTP) and some other
features of WebRTC (ICE, BUNDLE, rtcp-mux, ...), all of which rtpengine
can do.
Cheers
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