19 May
2016
19 May
'16
6:22 a.m.
What codecs are supported by your grandstream? Isn't the g711 in the group?
Cheers,
Daniel
On 19/05/16 01:51, Moacir Ferreira wrote:
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
--
Daniel-Constantin Mierla
http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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:
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 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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