Hi,
Thank you very much for the answers! I'll have a play.
Cheers, Yufei
Message: 9 Date: Mon, 8 May 2017 01:01:51 -0700 From: Maxim Sobolev sobomax@sippysoft.com To: sr-users@lists.kamailio.org, Daniel-Constantin Mierla miconda@gmail.com Subject: Re: [SR-Users] Can rtpproxy stream audio inside a call? Message-ID: CAH7qZfs0tQTwa-aCwvHHhAS_hajCRNaKBLnjXBv+O0txJNd4Mw@mail.gmail.com Content-Type: text/plain; charset="utf-8"
Daniel is right. Technically speaking it can be done at any point after the media session has been established, but you need some kind of trigger to start/stop it from your routing script. Such as re-INVITE for example.
-Max
On May 8, 2017 7:52 AM, "Daniel-Constantin Mierla" miconda@gmail.com wrote:
Hello,
it can play rtp files encoded in the format expected by the audio codec. I used only when there was a re-invite (like putting the call on hold).
Cheers, Daniel
On 04.05.17 17:29, Yufei Tao wrote:
Hi,
Just a quick question: if I use the Kamailio rtpproxy module with the
Sippy
RTPproxy, can I stream audio within a call? Or is it only possible before call audio is established?
For example can I mix/playback audio file into an existing established call session at any time using functions rtpproxy_stream2uac() and
rtpproxy_stream2uas()
somehow? If it is possible, does it need to be triggered by an in-dialog message from either UAC or UAS?
Cheers, Yufei
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Yufei Tao