Codec Translation path

Asterisk Asterisk Support
1

Hi All
I am new to Asterisk,I am using CentOS 5.9 final version at server and client end, I am sending voip traffic from asterisk 1.8 server through openvpn to asterisk “1.8” clients.
These codecs are installed " g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g719 speex16 testlaw".
There are GOIP 16 ports GSM gateways with asterisk client pc
When I send traffic from server to clients it works fine with 5 calls but when I increase the number of calls, call fails. I need to change codec translation, but not know from where I can change it.

I want to run 64 ports on 512kbps,
[color=#FF0000]

core show translation
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g719 speex16 testlaw
g723 - 2000 1001 1001 1999 1001 1000 2000 3000 3999 4999 3000 1001 - - 1002 - 3002 1001
gsm 2000 - 2 2 1000 2 1 1001 2001 3000 4000 2001 2 - - 3 - 2003 2
ulaw 2000 1001 - 1 1000 2 1 1001 2001 3000 4000 2001 2 - - 3 - 2003 2
alaw 2000 1001 1 - 1000 2 1 1001 2001 3000 4000 2001 2 - - 3 - 2003 2
g726aal2 2000 1001 2 2 - 2 1 1001 2001 3000 4000 2001 2 - - 3 - 2003 2
adpcm 2000 1001 2 2 1000 - 1 1001 2001 3000 4000 2001 2 - - 3 - 2003 2
slin 1999 1000 1 1 999 1 - 1000 2000 2999 3999 2000 1 - - 2 - 2002 1
lpc10 2999 2000 1001 1001 1999 1001 1000 - 3000 3999 4999 3000 1001 - - 1002 - 3002 1001
g729 2999 2000 1001 1001 1999 1001 1000 2000 - 3999 4999 3000 1001 - - 1002 - 3002 1001
speex 2999 2000 1001 1001 1999 1001 1000 2000 3000 - 4999 3000 1001 - - 1002 - 3002 1001
ilbc 2999 2000 1001 1001 1999 1001 1000 2000 3000 3999 - 3000 1001 - - 1002 - 3002 1001
g726 2999 2000 1001 1001 1999 1001 1000 2000 3000 3999 4999 - 1001 - - 1002 - 3002 1001
g722 2000 1001 2 2 1000 2 1 1001 2001 3000 4000 2001 - - - 1 - 2001 2
siren7 - - - - - - - - - - - - - - - - - - -
siren14 - - - - - - - - - - - - - - - - - - -
slin16 2001 1002 3 3 1001 3 2 1002 2002 3001 4001 2002 1 - - - - 2000 3
g719 - - - - - - - - - - - - - - - - - - -
speex16 3001 2002 1003 1003 2001 1003 1002 2002 3002 4001 5001 3002 1001 - - 1000 - - 1003
testlaw 2000 1001 2 2 1000 2 1 1001 2001 3000 4000 2001 2 - - 3 - 2003 -
[/color]

I want to change it with this out put which is as under

[color=#00BF00]
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g719 speex16 testlaw
g723 - 2000 2 2 1001 2 1 1001 2001 4000 4001 2001 2 - - 3 - 2003 2
gsm 3001 - 2 2 1001 2 1 1001 2001 4000 4001 2001 2 - - 3 - 2003 2
ulaw 3001 2000 - 1 1001 2 1 1001 2001 4000 4001 2001 2 - - 3 - 2003 2
alaw 3001 2000 1 - 1001 2 1 1001 2001 4000 4001 2001 2 - - 3 - 2003 2
g726aal2 4000 2999 1001 1001 - 1001 1000 2000 3000 4999 5000 3000 1001 - - 1002 - 3002 1001
adpcm 3001 2000 2 2 1001 - 1 1001 2001 4000 4001 2001 2 - - 3 - 2003 2
slin 3000 1999 1 1 1000 1 - 1000 2000 3999 4000 2000 1 - - 2 - 2002 1
lpc10 4000 2999 1001 1001 2000 1001 1000 - 3000 4999 5000 3000 1001 - - 1002 - 3002 1001
g729 4000 2999 1001 1001 2000 1001 1000 2000 - 4999 5000 3000 1001 - - 1002 - 3002 1001
speex 3001 2000 2 2 1001 2 1 1001 2001 - 4001 2001 2 - - 3 - 2003 2
ilbc 4000 2999 1001 1001 2000 1001 1000 2000 3000 4999 - 3000 1001 - - 1002 - 3002 1001
g726 3001 2000 2 2 1001 2 1 1001 2001 4000 4001 - 2 - - 3 - 2003 2
g722 3001 2000 2 2 1001 2 1 1001 2001 4000 4001 2001 - - - 1 - 2001 2
siren7 - - - - - - - - - - - - - - - - - - -
siren14 - - - - - - - - - - - - - - - - - - -
slin16 4001 3000 1002 1002 2001 1002 1001 2001 3001 5000 5001 3001 1000 - - - - 2000 1002
g719 - - - - - - - - - - - - - - - - - - -
speex16 4002 3001 1003 1003 2002 1003 1002 2002 3002 5001 5002 3002 1001 - - 1 - - 1003
testlaw 3001 2000 2 2 1001 2 1 1001 2001 4000 4001 2001 2 - - 3 - 2003 -

[/color]

Kindly guide me from where I can change it to get the above chart output

regards
kashif imran

2

Not (sensibly) possible, to change the table.

I doubt that 8kbps per channel is possible except with special compression and multiplexing techniques, as the overhead for simple RTP with standard packetisation intervals is larger than that. I would start by eliminating all the codecs that require more than 6kpbs, for the raw media, from the allow list. I’m not sure if Asterisk actually has any such codecs. Then you will have to solve the problem of mulltiplexing them efficiently.

IAX may be better than SIP.

I believe there is third part software, but that is generally only used in low wage countries.

For anyone else reading this, whilst the codec translation paths table is about CPU usage, I think the resource limit here is network bandwidth. The first line for controlling that is the allow rules, but that isn’t going to be enough.

3

Dear David
Yes you are right, I wanna use it for low bandwidth, I wanna run 64 channels on 512Kbps. I am using iax, my first server was configured by an asterisk professional, while I installed the 2nd server, the only difference which I found was codec which I past on this page, When I start traffic the server which I configured works fine only with five calls or you can say it supports only 5 simultaneous calls

this is the codec translation of the server which I configured

[color=#FF0000]gsm 2000 - 2 2 1000 2 1 1001 2001 3000 4000 2001 2 - - 3 - 2003 2[/color]

While this is the output of codec translation of the server which was configured by an asterisk professional, it supports 64 calls on 512 Kbps
[color=#00BF00]gsm 3001 - 2 2 1001 2 1 1001 2001 4000 4001 2001 2 - - 3 - 2003 2 [/color]

please compare the two tables u will get the difference, I need the out put in green color,

Kindly guide me what to do and where to do

Thanks
kashif

4

Hi
Core show translation output has Nothing to do with bandwidth, its shows the translation time between codecs.
If someone got 64 calls down a 512k pipe then get them to do it again. as from asteriskguru.com/tools/bandw … ulator.php you will see its not really possible.

5

hi
I have one Asterisk server with a big pipe of bandwidth, this server can support up to 120 simultaneous calls, so I send the traffic from server to various asterisk client PC each client pc “P4 1.8GHz processor, 2GB RAM” can support up to 64 calls with 1024Kbps upstream and 4088Kbps down stream DSL. I am already running this solution, I have configured asterisk Server and asterisk client pcs, the only thing which is being problem for me is the number of calls.

When I use my own configured server it support only five calls, if I use the server which is configured by a professional it supports 64 calls on one DSL. Thing I wanna know is how can I change the codec rate.

The only difference in these two server is codec rate, how and from where I can change it.

regards

6

Hi

READ my last post . core show translation does NOT reflect codec bandwidth its the translation time

Also there is a BIG difference between getting 64 calls up a 1 meg link to geting 64 up a 512k link.

you have posted NO debug output from calls on the server in question… if you can olny get 5 then i guess you are using alaw or ulaw

Just copt the configs from the other server and put them on yours.