[asterisk-users] My G729 problem re-visited

[Scott Moseman]

That was actually a VM.  Here's the real server (13ms).

CLI> show translation
          g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
     ulaw    -   3    -    1        2     2    1     3   13     -   15    2    -
     alaw    -   3    1    -        2     2    1     3   13     -   15    2    -
     g729    -   5    4    4        4     4    3     5    -     -   17    4    -

# dmesg | grep 'Xeon(TM)'
CPU0: Intel(R) Xeon(TM) CPU 2.80GHz stepping 03
CPU1: Intel(R) Xeon(TM) CPU 2.80GHz stepping 03

Thanks,
Scott


On 10/12/07, Mike Lynchfield <theclubvoip at gmail.com> wrote:
>
> 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
> >