[OpenSIPS-Users] Generating 183 reply and Playing Early Media using rtpproxy_stream2uac()

[Faheem Muhammad]

Husnain,
The same type of question is already answered by Kristian F. Høgh on the
opensips mailing list. Try to search "*[OpenSIPS-Users] Playing caller a
file before dialing callee*"

Hope it will solve your problem.

Faheem


On Thu, Jan 7, 2016 at 10:37 AM, Hamid Hashmi <hamid2kviii at hotmail.com>
wrote:

> Try the following example. Change connection IP and codec order
> accordingly.
>
> if (is_method("INVITE") && has_body("application/sdp")) {
>     $var(Session_owner) = $rb[1];
>     append_to_reply("Content-Type: application/sdp\r\nv=0\r\n$var(Session_owner)\r\ns=call\r\nc=IN IP4 10.130.130.114\r\nt=0 0\r\nm=audio 61896 RTP 0 8 3 101\r\na=rtpmap:0 pcmu/8000\r\na=rtpmap:8 pcma/8000\r\na=rtpmap:3 gsm/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n")';
>     t_reply_with_body("183", "Session Progress", "$var(body)");
> }
>
>
>
> *Hamid R. Hashmi*
> Software Engineer - VoIP
> Vopium A/S
>
>
> ------------------------------
> Date: Wed, 6 Jan 2016 20:33:29 +0300
> From: husnain.taseer at gmail.com
> To: users at lists.opensips.org
> Subject: [OpenSIPS-Users] Generating 183 reply and Playing Early Media
> using rtpproxy_stream2uac()
>
>
> Dear Users,
> I have a scenario where I want to Play an announcement as early media to
> the UAC before answering the call but I don't want to use any media server
> like asterisk/Freeswitch.
>
> When user agent sends an INVITE I am calling rtpproxy_offer() and sending
> INVITE to B party. On 100 Trying from B party I am
> calling rtpproxy_stream2uac() and streaming the file I can see that RTPs
> are going towards the UAC (caller) but softphone is not accepting those
> RTPs because 183 was not sent to the softphone so he don't know the media
> details of the rtpproxy. but as 200 Ok reaches to the softphone last part
> of the audio can be heard immediately after Answer.
>
> So I think on 100 Trying from B Part if I send 183 Session Progress to the
> softphone and then starting the RTP stream will work. So can you please
> tell me is there a way to generate 183 Session Progress with media details
> of RTPPROXY in opensips ? so that my scenario starts work.
>
> Regards,
> Husnain Taseer
> VoIP Developer
>
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