[Live-devel] playSIP crashed after getting 200 OK for invite

[Vilaysak Thiphavong]

Hi Ross,

Thank you for your advice

I am able to play out a Wav file (RTP payload)  without RTSP in to a connected SIP session and I have a few questions below:

1. What is different between OnDemandServerMediaSubsession and ServerMediaSubsession classes?

2. What is different between RTPSink and SimpleRTPSink classes?

3. Can I play/stream a wav file with g729 codec, which testProgram does this ? (so far I am successful with g711 (alaw/uLaw) and AMR )

4. Could you please address me to the class where I can record incoming RTP stream in to a wav file format?

 
Thanks,
Sak



-----Original Message-----
From: live-devel [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Vilaysak Thiphavong
Sent: 11 February 2016 13:32
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] playSIP crashed after getting 200 OK for invite


Hi,

I have a SIP application (written in C) and my RTP server also in C.  Below is how they work

>From SIP client side

1. Create a socket connection to RTP server using RTP client library 2. Then send CREATE_CMD to RTP server to create rtp media session with SDP parameters for this call 3. Send INVITE 4. After call is established send PLAY_CMD/RECORD_CMD to RTP server to play/record a wav file

Because there are some limitations within my RTP server and I would like to replace it with RTP module from Live555 package  

I am newbie in LIVE555 and also on learning edge of C++. 

Could anyone shed any light on this how can I make use of RTP module from Live555 package in my current project?
How can I by pass RTSP server and directly call RTP library?

Any help will be greatly appreciated. 

Thank you,

Sak
   




-----Original Message-----
From: live-devel [mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson
Sent: 08 February 2016 15:02
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] playSIP crashed after getting 200 OK for invite

Thanks for the note.

I’ve just released a new version (2016.02.08) of the code that should fix this.


Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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