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[Linphone-developers] Status of support of communication with WebRTC age

From: Andrey Utkin
Subject: [Linphone-developers] Status of support of communication with WebRTC agents
Date: Mon, 8 Dec 2014 17:06:56 +0200

Date: Mon, 8 Dec 2014 17:03:58 +0200
Subject: RTP/SAVPF support status
Hi Linphone developers!

We are interested in direct communication of webrtc-based VoIP agents
and LinphoneAndroid-based app. But as I see this doesn't work:
- in the call in webrtc -> linphone-android direction, because
linphone replies "488 Not Acceptable", I guess it is because of
RTP/SAVPF in SDP from webrtc.
- in linphone-android -> webrtc direction the call also gets rejected
(didn't track the secure web socket traffic to figure out what exactly
is the response from browser).
I assume usage of Freeswitch or Asterisk.
Webrtc2sip broker usage is not an option, this software is
unmaintained and doesn't work correctly and stable.

Could anybody please comment on the status of such interconnection capabilities?
- is there an easy way to interconnect webrtc and linphone-android
agents seamlessly without complete media streams reencoding?
- what is lacking?
- for what is lacking, how much efforts must be put, and what are
estimated costs?

Thanks in advance for any info.

Andrey Utkin

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