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
