Hi all, We would like to increase the size (sample length) of RTP packets sent by Asterisk to SIP phones. I gather that Asterisk currently always uses 20ms packets for RTP, although I can't find in the source where that's defined, unless it's in chan_zap.c.
I'm guessing from [http://lists.digium.com/pipermail/asterisk-users/2003-September/019490.html] that it's not possible right now (or at least wasn't in September). Does anyone know if and when it's coming as an option, or if we can modify the source to always use, say, 40ms packets instead of 20? Thanks in advance for any advice. Please CC me since I'm not subscribed to the list anymore (too much traffic). Cheers, Chris. -- _ __ __ _ / __/ / ,__(_)_ | Chris Wilson -- UNIX Firewall Lead Developer | / (_ ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk | \__/_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 | _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
