Hi guys,
I have a weird problem, and I have encountered a few other people with the same issue. The problem is this:
Whenever I make a call from my IAXy (g711ulaw) to my server, and then my server transcodes to speex and sends it to a remote asterisk server, audio is perfectly fine. The same goes if I use Linphone with speex. However, whenever I use a Cisco 7960 SIP 6.2 or a Polycom IP 500 with SIP 1.3.1.0056, both using g711ulaw to my server, when it is transcoded to speex and a connection is made to the remote system, audio chops and I lose about 80% of all audio. This has been tested on asterisk 1.0.3, 1.0.4, 1.0.5, and CVS v1-0 from yesterday, all operate identically. The problem works identically when I go to the remote and call to my server with it transcoding from the Polycoms to my server.
Hardware & software:
My server: Dual Proc Xeon 2.8GHz/800Mhz FSB, 1 GB ECC Reg RAM, Intel SE7525GP2 motherboard, X101P (digium), 3ware RAID controller 9500. Slackware 10.0, speex 1.0.4, libogg-1.1-i486-1, libvorbis-1.0.1-i486-1.
Remote server: HP Proliant ML330 Xeon 3.06Ghz, Smartarray SCSI RAID, 512MB ECC Reg RAM, TDM04b (digium). Slackware 10.0, speex 1.0.4, libogg-1.1-i486-1, libvorbis-1.0.1-i486-1.
Translation times are shown in asterisk on both machines as being in the vicinity of 28 to 45 ms from all other codecs to speex.
Ping times between machines: 24 packets transmitted, 24 received, 0% packet loss, time 23230ms rtt min/avg/max/mdev = 15.770/53.820/164.989/43.473 ms
My codecs.conf: [speex] ;0-10 quality => 3 ;0-10 complexity => 4 ; true / false enhancement => true ; true / false vad => false ; true / false vbr => false ;0-10 abr_quality => 5 ; true / false abr => false ;0-10 vbr_quality => 5 ; true / false dtx => false
If anybody has any insight to this problem, it would be appreciated. (BTW, Speex 1.1.6 is just that, unstable... it will even crash asterisk when you try to do "show translation recalc").
Thanks, -Mishehu _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
