[asterisk-users] My G729 problem re-visited

[Mike Lynchfield]

How do you get 11ms translation time on ulaw 729 ?

we have 12ms and its dual xeons 2.6..

On 9/26/07, Scott Moseman <scmoseman at gmail.com> wrote:
>
> Ok, I built a test system to duplicate my problem and provide myself
> a platform that I can mess around with to try and break any features.
> My problem is G729 pass-through from a gateway to a phone. I think
> I even have transcoding working, which makes me more confused on
> what's wrong with my pass-through. It must be a configuration issue.
>
> The basics...
>
> *CLI> core show version
> Asterisk 1.4.11 built by root @ fwd-tst02 on a i686 running Linux
>
> *CLI> show modules like 723
> Module Description Use Count
> codec_g723.so G.723.1 Coder/Decoder 0
> format_g723.so G.723.1 Simple Timestamp File Format 0
>
> *CLI> show modules like 729
> Module Description Use Count
> codec_g729.so G.729 Coder/Decoder 0
> format_g729.so Raw G729 data 0
>
> *CLI> show translation
> [truncated]
> g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
> ulaw 5 2 - 1 2 2 1 3 7 - 11 2 -
> alaw 5 2 1 - 2 2 1 3 7 - 11 2 -
> g729 5 2 2 2 2 2 1 3 - - 11 2 -
>
> The configuration...
>
> [gateway]
> type=friend
> host=gateway
> context=default-inbound
> disallow=all
> allow=g729
>
> [phone]
> type=friend
> context=sip
> host=dynamic
> username=phone
> secret=scott
> dtmfmode=RFC2833
> disallow=all
> allow=g729
> callerid=Scott
> qualify=yes
> canreinvite=no
>
> exten => 1266,1,Dial(SIP/[number],30,t)
> exten => 1266,2,Congestion
>
> exten => 1266,1,Dial(SIP/[number],30)
> exten => 1266,2,Congestion
>
> (The same results using both of the above dialplans...)
>
> The environment...
>
> PSTN -> Gateway -> Asterisk -> Phone
>
> What I'm seeing works...
>
> With the gateway setup to send both G711 and G729, it sends
> an INVITE which includes both G711 and G729 codecs. Asterisk
> sends an INVITE to my phone with only G729. The call is made
> and there's a conversation in G711 with the gateway and G729
> with the phone. I assume this means Asterisk is transcoding.
>
> What I"m seeing fails...
>
> With the gateway setup to send only G729, it sends an INVITE
> to Asterisk which includes only G729. Asterisk send an INVITE
> to the phone using G729, too. The 200 OK from the phone to
> the Asterisk includes G729. The 200 OK going from Asterisk to
> the gateway doesn't include ANY codec. The call is dropped the
> moment I pickup the phone to answer the call.
>
> My question...
>
> Why does Asterisk not want to respond to my gateway in G729?
> Even if the gateway requests it, Asterisk seems to just ignore it.
> From the transcoding call, and phone to phone G729 calls, I have
> proof that Asterisk knows how to handle G729 calls.
>
> Where do I go from here???
>
> Thanks,
> Scott
>
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-- 
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
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