Steve- > Sorry for not replying sooner. It turns out your ideas were helpful. Searching > for ptime actually did find a location where I was able to set the frames per > packet. Ok sounds good. -Jeff > On Mon, Mar 15, 2010 at 9:12 PM, Jeff Brower <jbrower at signalogic.com> wrote: > > Steven- > > > > Thanks for the reply, but you have not addressed my question, which is: > > > > how do I change the bytes per RTP packet? > > > > I am aware of the consequences. I just need to know how to do it. > > > I don't know how, my thought was if you looked for anything related to > "vif" you could get some clues. Also suggest to search pjsip.org for > "ptime" -- seems to be packet payload time in msec. > > Sorry I'm not able to be more helpful. > > -Jeff > > > > On Mon, Mar 15, 2010 at 6:51 PM, Jeff Brower <jbrower at signalogic.com> > > wrote: > > > > Steven- > > > I set the config to use the GSM codec. According to > > various web sites, GSM > > > should be 13 Kbps, but I'm measuring around 35 Kbps. I > > understand that this > > > difference is caused by overhead from bytes added due to > > the RTP, UDP and IP > > > packets. > > > > > > I also see that each RTP packet contains 33 bytes of GSM > > data while the > > > entire packet is 87 bytes. That's over 160% overhead. > > > > > > How can I change the frame size to include more GSM data > > bytes in each RTP > > > packet? I would like to increase the size of each packet > > to around 165 (5 > > > times the current size). This would decrease the overhead > > to about 32%. I > > > realize this will also increase the latency 5 times as > > well. > > > > > > So how can I change the bytes per frame for the GSM > > codec?In traditional VoIP software this is called "VIF size" > > (voice information field), a way to specify integer multiples > > > > of the codec's normal packet size. Not sure what it's called > > in pjsip. > > > > But the other end has to be able to handle this, so something > > to worry about. Also 5x the normal GSM packet will be > > 100 msec latency, which is a lot. You might exceed > > capabilities of carrier's echo cancellers depending on what > > is the > > remote endpoint. > > > > -Jeff > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20100326/c77c57d8/attachment.html>
