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/20100315/e9e410c3/attachment.html>
