[Freeswitch-users] RTMP + G.729 Codec

[x.liu]

Hi,

I am also trying RTMP calls. I am using the Flex calling to Nunace MRCP 
server for speech recognitions.
The same setup works well via the VoIP phone calls but not works via a 
web browser calls (via adobe Flex).
I notice there are a lot of "audio queue overflow" debugging outputs 
when calling via the Flex

I am wondering if any body has ever tried and get it working?

Thanks!

Xing


On 10/03/2012 05:01 PM, Balamurugan Mahendran wrote:
> Thanks!!
>
> --Bala
>
> On Wed, Oct 3, 2012 at 9:23 PM, Steven Ayre <steveayre at gmail.com 
> <mailto:steveayre at gmail.com>> wrote:
>
>     G729 is not supported by the RTMP protocol.
>
>     See page 14 of
>     http://wwwimages.adobe.com/www.adobe.com/content/dam/Adobe/en/devnet/rtmp/pdf/rtmp_specification_1.0.pdf
>
>     Speex is the one used by default.
>
>     -Steve
>
>
>
>
>     On 1 October 2012 15:12, Balamurugan Mahendran <b2m at a-cti.com
>     <mailto:b2m at a-cti.com>> wrote:
>     > I am working on rtmp call quality, Is it possible to use g.729
>     for rtmp? and
>     > will it increase the quality?
>     >
>     > Now I am experiencing Echo + large delay and background noise.
>     >
>     > Thanks,
>     > Bala
>     >
>     >
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