19 May
2016
19 May
'16
1:51 a.m.
I did not dig into the problem but on my tests I saw that my (old) Grandstream phone was
refusing the call for not having a compatible codec to talk with the offered ones by the
browser (Firefox). Being this the case, I guess I must include a translator, and all
routing logic, in between the callers. It points to Asterisk that I would like to avoid
for now. But I guess this is not a problem that only affects me. Someone else must have
faced this before. So the question still open: What solution would be recommended for such
case?
Cheers,
Moacir
To: sr-users(a)lists.sip-router.org
From: rfuchs(a)sipwise.com
Date: Wed, 18 May 2016 19:03:10 -0400
Subject: Re: [SR-Users] Browser WebRTC transcoder
On 18/05/16 04:57 PM, Moacir Ferreira wrote:
Hey Daniel,
If you say so, you probably right... I did not try it because on the
sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:
/"Rtpengine does not (yet) support:/
//
* /Repacketization or transcoding/
This refers to translating one audio codec into another (e.g. opus to
PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting)
is supported.
Cheers
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