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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