[Asterisk-Users] Speex transcoding for Cisco / Polycom

I put the Who? in Mishehu mishehu at shavedgoats.net
Fri Feb 25 09:54:54 MST 2005


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



More information about the asterisk-users mailing list