[asterisk-users] Asterisk 1.4 and g723

[Andrew Joakimsen]

What G723 codec do you have on Asterisk? What is your SIP.CONF? What
ATA/Phone is being used and what are the exact settings, especially
for the codec?

On 1/19/07, Phil French <pfrench at caprock.com> wrote:
> I am setting up Asterisk for use in a low bandwidth environment.  As
> bandwidth is precious and our ATA's support it, the decision was made to
> use the g723 codec.  I have been working on this for a few days and have
> not been successful.  The issue that I am having is garbled noise at the
> client on calls whose RTP streams are terminated by Asterisk system.
> This is the case for all the dialplan applications I have tested except
> for Echo.  The critical application for us is Voicemail.  When a call to
> voicemail extension is initiated the Asterisk console does not indicate
> any error.  Packet captures indicate the call is active and streaming
> g723 data.  Everything seems well but is not.  The audio at the client
> is unrecognizable.  One thing that I have noticed is that the bitrates
> in the upstream and downstream direction differ.  From Asterisk to ATA
> the g723 bitrate switches between 5.3 kb/s and 6.3 kb/s.  From ATA to
> Asterisk the bitrate is a constant 6.3 kb/s.  I don't think this is a
> problem but seems odd.  As a comparison I captured packets from a call
> to the echo application and found that the bitrate was 6.3 kb/s in both
> upstream and downstream packets.  Additionally, all prompts are g723
> format.  Voicemail is saved as g723sf.  As a parrallel task a co-worker
> has gotten 1.2 to work with g723.  However we require 1.4 for t.38
> pass-through.
>
> The end-to-end system is illustrated below.
>
>                       [Asterisk]
>                        /     \
>                      ip       ip
>                      /         \
>   [PSTN]--pri--[GATEWAY]--ip--[ATA]<--2pr-->[Phone]
>
> System details
>  -Asterisk server version 1.4 - compiled from source - Fedora Core 6
> -Gateway - Cisco 2811  -ATA - Linksys 2102
>
> I would appreciate any advice or suggestions.  It should be noted that
> the calls to the PSTN through the gateway and calls between ATA's are
> working fine.
>
> Regards,
>
> Phil French
>
> Phil French
> Systems Engineer
> -------------------------------
> CapRock Communications
> 4400 S. Sam Houston Parkway E.
> Houston, Texas 77048
> Office: 832 668 2643
> pfrench at caprock.com
> www.caprock.com
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