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
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
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
On dinsdag 12 april 2016 11:42:53 CEST Alex Lutay wrote:
> The point here that Kamailio checks DB table_version and doesn't start
> if the version mismatch found.
As long as table versions are backwards compatible, just use a separate
"version" table for each Kamailio version you deploy. Set the
Hi Daniel,
I know its been some time since looking at this but I finally had the time to
dig deeper. It appears that the timestamps inserted by the sip trace module
are in host order, not network order. This obviously creates an issue when the
endianess of the agent and capture server do not
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 transcodingPlayback of pre-recorded
streams/announcementsRecording of media streamsZRTP"
Hello,
kamailio+rtpengine should do this job quite well.
Cheers,
Daniel
On 18/05/16 19:16, Moacir Ferreira wrote:
> A question for the community:
>
> What would be your best advice for a RTP proxy/transcoder to allow
> browser WebRTC calls to legacy VoIP?
>
> Moacir
>
>
>
A question for the community:
What would be your best advice for a RTP proxy/transcoder to allow browser
WebRTC calls to legacy VoIP?
Moacir
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
Great Daniel! Problem solved.
Thanks,
Moacir
To: sr-users@lists.sip-router.org
From: mico...@gmail.com
Date: Wed, 18 May 2016 07:08:02 +0200
Subject: Re: [SR-Users] WebRTC
Hello,
if you don't have a trusted certificate, then browse first to
https://kamailioip:5061
