Hi Kabil; Could you explain more about what you are trying to do.. I mean as a first look at your mail i suspect that you are trying to build an rtp stream from one point to another ... you may use only rtp libraries such as ortp .. Does sip involve with your application ? May be it is better to think about what you really need? Modifying pjsips rtp and media management would not be an answer for your application since pjsips rtp and audio media bounds are really tight as far as i see.. regards muge 2010/10/1 Kabil Akp?nar <kabilakpinar at gmail.com> > So anybody? > > 2010/10/1 Kabil Akp?nar <kabilakpinar at gmail.com> > > Hi all; >> I am trying to give m=application support in pjsip, and close to the end. >> But before I proceed, I need to double check the "stream" issue. >> For m=audio stream has encoder and decoder attached and a jitter buffer. >> In m=application case, incoming data does not need any encoder or decoder >> for the moment, so incoming/outgoing must just pass without any >> modification. In this case I need to either remove enc/dec parts for >> m=application and modify get_frame/put_frame or I should introduce a dummy >> encoder/decoder, but removing seems easier, latter is not that hard in fact. >> Also the buffer used for storing incoming/outgoing data is not necessary, >> simply the data can be immediately passed to upper level by callback >> mechanism.. >> >> So what would be your very valuable suggestions? >> >> -- >> Kabil Akp?nar >> >> > > > -- > Kabil Akp?nar > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20101001/1f2904ca/attachment-0001.html>
