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
To: sr-users@lists.sip-router.org From: rfuchs@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
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users