[streaming] RTP streaming - codecs question

[Diego Susa]

Hi
I'm re-posting this cause it was grouped in a "not-intended" thread.
I'm new to VideoLAN and streaming, but I'm trying to see if it fits in
my project.
What I'm trying to do is using the rtsp_VoD module to feed video
and audio clips into some SIP softphones via RTP. 

Well, the video streams are not a problem because there's not so
much to choose. As far as I know H263 seems to be the most widespread
codec, and H264 the "next to be"...

But the situation is not that clear with the audio codecs.
The most common Audio codecs (and their payload types) in softphones
are: PCMU/8000 (pt:0), GSM/8000 (pt:3) and G723/8000 (pt:4). There also
some others very popular like G729, DVI4, and Speex.
The problem is that "none" of them seem to be supported at the RTP
output_stream module of VLC
... or am I missing something ?

I've searched through /modules/stream_out/rtp.c, and I've just found
these functions:
rtp_packetize_l16;
rtp_packetize_l8;
rtp_packetize_mpa;
rtp_packetize_mpv;
rtp_packetize_ac3;
rtp_packetize_h263;
rtp_packetize_split;
rtp_packetize_mp4a;
rtp_packetize_mp4a_latm;
rtp_packetize_amr;
rtp_packetize_amr;

And NONE of them account for any of the audio codecs I've mentioned
above ...  :-( 
Does it mean I should give up and asume that there's no way to send
those audio streams using RTP ?

In the case of the "sad" answer. Please give me a hint on this:
What I've seen at that "rtp.c", and also at "rtsp.c", is that the
streams are evaluated by their FOUR-CC codes, in order to choose the
right "packetizer". When I load some simple audio files that have
FOURCC="ulaw", the switch/case fails to find a match and tells me
"unssuported codec". Does this suggest that I could "add" support
for other codecs, by adding a new "case" and a new "packetizer"
function, for example for a PCMU/8000 payload ? 
... in the case I could find out how to build that function, of course  :-) 

thanks in advance for any help,
cheers !

Diego Susa



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