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[asterisk-users] Asterisk 13 and WebRTC

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annusfictus at gmail.com
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Posted: Thu Sep 08, 2016 4:37 pm Post subject: [asterisk-users] Asterisk 13 and WebRTC

Hello list, before to lost my time, I'd like know if someone have a WebRTC working configuration on Asterisk 13.11.0 SIP or PJSIP channel. Thank you Regards -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

ovoshlook at gmail.com
Guest

Posted: Fri Sep 09, 2016 2:34 am Post subject: [asterisk-users] Asterisk 13 and WebRTC

Hi. It have big audio delay because using extenral ICE servers.
Better to use kamailio/opensips + rpenigne infront

2016-09-09 0:36 GMT+03:00 Annus Fictus <annusfictus@gmail.com (annusfictus@gmail.com)>:

Quote:

Hello list,

before to lost my time, I'd like know if someone have a WebRTC working configuration on Asterisk 13.11.0 SIP or PJSIP channel.

Thank you

Regards

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
Â Â Â http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
Â Â Â https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Â http://lists.digium.com/mailman/listinfo/asterisk-users

cervajs2 at gmail.com
Guest

Posted: Fri Sep 09, 2016 3:47 am Post subject: [asterisk-users] Asterisk 13 and WebRTC

using in production

last asterisk 13 + pjsip bundled + pjsip patch for RTP/SAVPF (search
pjsip conf) + sipml5 version from roginvs

https://github.com/DoubangoTelecom/sipml5/pull/238

Dne 08/09/2016 v 23:36 Annus Fictus napsal(a):

Quote:

Hello list,

before to lost my time, I'd like know if someone have a WebRTC working
configuration on Asterisk 13.11.0 SIP or PJSIP channel.

Thank you

Regards

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

annusfictus at gmail.com
Guest

Posted: Fri Sep 09, 2016 7:56 am Post subject: [asterisk-users] Asterisk 13 and WebRTC

Hello,

I mean a working configuration (SIP o PJSIP) without patches or code
corrections.

Thank you

Regards

El 09/09/2016 a las 03:47, marek cervenka escribió:

Quote:

Hello list,

before to lost my time, I'd like know if someone have a WebRTC
working configuration on Asterisk 13.11.0 SIP or PJSIP channel.

Thank you

Regards

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