[asterisk-users] G.729 for Asterisk

Gordon Henderson gordon+asterisk at drogon.net
Tue Sep 15 05:55:53 CDT 2009


On Tue, 15 Sep 2009, Steve Totaro wrote:

> On Tue, Sep 15, 2009 at 5:09 AM, Gordon Henderson <
> gordon+asterisk at drogon.net <gordon%2Basterisk at drogon.net>> wrote:
>
>> On Tue, 15 Sep 2009, Steve Totaro wrote:
>>
>>> Asterisk supports this codec in passthrough without buying anything and
>> no
>>> real hit on the CPU since no transcoding necessarily takes place.
>>>
>>> A world wide Asterisk system/network using all G729 from phones, to make
>>> menuselect and selecting G729 sounds and getting a G729
>>> origniation/termination is a beautiful thing.
>>
>> Apart from MixMonitor, MeetMe and voicemail )-:
>>
>> And calls to GSM mobile phones where the transcode from G729 to GSM (via
>> alaw in the PSTN) doesn't sound that good - even to me and I have rubbish
>> ears...
>>
>> Gordon
>>
>> True enough,
>
> Luckily, I don't use mixmonitor in that deployment.  voicemails are few and
> far between in relation to call volume and meetme is only used regularly but
> with very few participants.  3 way calling is usually sufficient.

Indeed. An I use g729 end to the PSTN plumbing where the transcode is done 
by someone else, so it's good in that respect. An issue I have is the low 
power PBXs I like to make & use for local voicemail though...

> Would Nconference or whatever the drop in replacement to Meetme have G729
> issues.

No idea - I'm under the impression (from earlier messages here) that you 
simply can't mix a compressed audio stream without uncompressing it 
first...

> I guess the G729 to GSM depends on the provider.  Using L3 or XO, I have
> never really heard any issues.

I've had a few compliants - mostly from the people at the far-end on the 
mobile... (cell, handy, whatever you want to call it)

> On a double VSAT hop around the world with 10:1 or 20:1 contention on a
> small pipe, yeah, I have heard audio problems but I know for a fact that
> G729 is actually helping, not hurting.
>
> BTW, I do all SIP.

I used to use all IAX, but an migrating to SIP and finally going to dump 
IAX on back-hauls shortly. (Probably still use it for office to office 
though)

Gordon



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