<html><body bgcolor="#FFFFFF"><div>AFAIK, a rtp packet *is* a frame, so no you can't.</div><div><br></div><div>As the other poster said, use a higher ptime. That will make the frame store a longer period of time which will do the same as that you want.</div><div><br></div><div>Overhead will be lower, but quality will be worse if you drop packets or they fail to arrive in a timely manner. That's very likely on gsm, and the packet size will mean it takes longer to be delivered which'll mean more delay and probably more packets arriving too late. You'll need to experiment to find a good balance. And remember what works well in a city might fail to work in the country where coverage is poorer.<br><br>Steve on iPhone</div><div><br>On 18 Apr 2011, at 09:49, Kenneth Taylor <<a href="mailto:ktaylor91@yahoo.com">ktaylor91@yahoo.com</a>> wrote:<br><br></div><div></div><blockquote type="cite"><div><div style="font-family:'times new roman', 'new york', times, serif;font-size:12pt"><div>Hi,</div><div>I'm building a voip client for android phones over gsm and consider using FS as sever.</div><div>In order to save cellular costs I want to put more than one frame per Rtp packet.</div><div><br></div><div>Is there any easy solution to do it in FS? </div><div><br></div><div>Or, where is the right place to add it? In mod_sofia or sofia_glue? should I go to lower levels?</div><div>I don't think I should do it in the codec, it may cause problems in the rtp sequence number and probably a lot more.</div><div><br></div><div>TNX,</div><div>Ken</div><div style="position:fixed"></div>
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