Speex dev - Nov 2004 - jitter buffer

Hi Farhan,

It is interesting that GPRS is 9.6 kps, as per GPRS standard it should
be 115kps ? if it is 9.6kps is true then I would use speex codec on GPRS
for my project, since in some country they charge GPRS on flat rate
rather than expensive airtime.

Thanks for your work !

Regards

Danny 




-----Original Message-----
From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On
Behalf Of Ashhar Farhan
Sent: Tuesday, November 16, 2004 8:25 PM
To: speex
Subject: [Speex-dev] jitter buffer


jitter varioues from application to application and network to network.
let me give you an instance from my own work.

i am using speex in a voip application that has to work on gprs 
connection. the gprs gives just 9.6 kpbs up. hence, i have an upper
limit 
of 6kpbs on the codec and not more than 5 udp packets per sec. that
means, 
putting in 10 speex frames in every rtp packet.

i need to jitter compensate for about 50 msec, but that necessarily
means 
holding back 10 frames in the buffer before the next packet arrives.
oth, 
the desktop version will have to dynamically track the settle on rtp 
packet size by retraining itself with the incoming stream.

i needed info on two topics:

1) how can i use the VAD at an application level? i would like to make
my 
voip application send packets only when the user speaks and at that
point 
i would like to cut off my playback to avoid echo (voice operated 
transmit/receive changeover).

2) can i control the echo delay ?

- farhan
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Danny Chan wrote:
>Hi Farhan,
>
>It is interesting that GPRS is 9.6 kps, as per GPRS standard it should
>be 115kps ? if it is 9.6kps is true then I would use speex codec on GPRS
>for my project, since in some country they charge GPRS on flat rate
>rather than expensive airtime.
>
>  
>
GPRS can be configured in multiples of 9.6Kbps at the discretion of the 
network provider.  For example, T-Mobile in the US provides 4 timeslots 
downstream and 1 slot upstream.  So voice on this would have to fit into 
9.6Kbps to fit in the upstream channel. 

Another big problem with GPRS wrtp VOIP is that its latency is really 
high.  I measured average 1 second for a 1 way trip across the GPRS 
network here.  I believe this is caused by the block error correction 
scheme that they use on the data channel -- quite a few bits are encoded 
together leading to high latency.  The phone companies just didn't 
understand that there are higher layer protocols for handling data 
integrity.

Jamey