Does anybody knows if it's posible or if there is some develoment in course to be able to use longer transmit packet sizes (as long as I know this is fixed in 20ms now) with the compressed voip codecs in asterisk (g729, g726, gsm, etc). I need to use asterisk to connect remote sip clients with 24kb bandwidth lines and I'm using a licences g729 codec but because I can't increase the packet size to 40 or 60 ms in asterisk the connection is useless. Thanks very much Luis
Luis Vazquez [luis@teledata.com.uy] wrote:> Does anybody knows if it's posible or if there is some develoment in > course to be able to use longer transmit packet sizes (as long as I know > this is fixed in 20ms now) with the compressed voip codecs in asterisk > (g729, g726, gsm, etc). I need to use asterisk to connect remote sip > clients with 24kb bandwidth lines and I'm using a licences g729 codec but > because I can't increase the packet size to 40 or 60 ms in asterisk the > connection is useless. Thanks very much Luis >It wouldn't help you if there was an easy Asterisk patch for this, as the G.729 code is closed source and is therefore un-patchable. You could try SpeeX or LPC10 - or a 56k modem. :-) -- _/ _/ _/_/_/_/ _/ _/ _/_/_/ _/ _/ _/_/_/ _/_/ _/ _/ _/ _/_/ _/ K e v i n W a l s h _/ _/ _/ _/ _/ _/ _/ _/_/ kevin@cursor.biz _/ _/ _/_/_/_/ _/ _/_/_/ _/ _/
Luis Vazquez wrote:> Does anybody knows if it's posible or if there is some develoment in > course to be able to use longer transmit packet sizes (as long as I know > this is fixed in 20ms now) with the compressed voip codecs in asterisk > (g729, g726, gsm, etc). > I need to use asterisk to connect remote sip clients with 24kb bandwidth > lines and I'm using a licences g729 codec but because I can't increase > the packet size to 40 or 60 ms in asterisk the connection is useless.The quick and dirty way: ------------------------ In rtp.c, function "ast_rtp_write", in the "switch" statement, "AST_FORMAT_G729A" case, change the smoother creation to something larger. E.g.: rtp->smoother = ast_smoother_new(40); Keep in mind that you must set this into something valid (45 obviously is not valid). Recompile and you should be fine. The right (but longer) way: --------------------------- The ability to packetize variable number of frames per RTP packet for various codecs should be configurable from within the rtp.conf file. This requires some coding of course. Currently, I don't have time available to do it, but I could do it as soon as I find some free time.> Thanks very much > Luis >Michael.