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[Linphone-developers] RTT in RTP parse


From: julien thai
Subject: [Linphone-developers] RTT in RTP parse
Date: Tue, 28 Jun 2005 10:37:38 +0200

Dear Simon
 
Sorry for send this email to you cos i tried to send it thousand times to address@hidden but i failed. Thank you to resend it to forum.
 
Cos i made some simple test RTT in RTP parse, after read the source code. I think the unit of RTT here is microsecnde? I only want to ask if it is true or not?
 
Thank you so much.
 
JT

 
2005/6/16, Simon Morlat <address@hidden>:
Hello,

You should declare MSCodecInfo object like it's done for every codec in
linphone. The purpose of this object is tell how to create the MSFilter that
do the compressing/decompressing job, but as you modificed audiostream.c this
is unused. Declaring a dummy MSCodecInfo with types "G723" will be enough to
make this part of code happy.
This part of code is just checking that we have MSFilter in the mediastreamer
able to handle each format described in the config file.

Simon

Le Jeudi 9 Juin 2005 05:48, Vanitha Ramaswami a écrit:
> Hi Simon,
>
> In G.723.1, i can run in either 5.3kbps or 6.3 kbps. I
> think both of them will have
> the same payloadtype ie. 4. Its not possible to call
> rtp_profile_set_payload right.?
>
>
> In order to hook to my external DSP codecs, to make
> the changes minimal i modified
> the audiostream.c file in the function
> audio_stream_start_full,
>
> ms_filter_add_link(stream->soundread,stream->rtpsend);
> ms_filter_add_link(stream->rtprecv,stream->soundwrite);
>
> With these changes it worked fine for me for the G.711
> MuLaw & A-Law, but now
> when i add support for G.723, it is giving me error
>
> in "coreapi/misc.c There is no plugin to handle.."..
>
> Do i need to stub out the G.723 encoder&decoder
> inorder to remove these kind of
> errors  from the medistreamer.
>
>
> I appreciate your help on this.
>
>
> Thanks
> Vanitha
>
> --- Simon Morlat <address@hidden> wrote:
> > Hello,
> >
> > I' ve added this support in cvs version.
> > It consists in adding this:
> >
> > PayloadType payload_type_g7231=
> > {
> >     TYPE( PAYLOAD_AUDIO_PACKETIZED),
> >     CLOCK_RATE(8000),
> >     BITS_PER_SAMPLE( 0),
> >     ZERO_PATTERN(NULL),
> >     PATTERN_LENGTH( 0),
> >     NORMAL_BITRATE( 6300),
> >     MIME_TYPE ("G723")
> > };
> >
> > in src/avprofile.c
> >
> > and add a
>
> rtp_profile_set_payload(&av_profile,4,&payload_type_g7231=);
>
> > somewhere in the av_profile_init() function in
> > src/avprofile.c
> >
> > Simon
> >
> > Le Lundi 6 Juin 2005 03:39, Vanitha Ramaswami a
> >
> > écrit:
> > > Hi Simon & others,
> > >
> > > I need to G.723 support to the oRTP stack. Is it
> > > enough that i define a new
> > > payloadtype in the file avprofile.c ? I have
> >
> > external
> >
> > > DSP chip that does the G.723 compression. Can you
> > > please let me know if need to do anyother
> > > changes.
> > >
> > > Thanks
> > > Vanitha
> > >
> > > __________________________________________________
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