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G.729 sample size

CHRIS HABBERJAM
Level 1
Level 1

I've a customer who records in G.711 format and most users use this, but wants to use 30ms sample size for playing out if G.729 is requested by the remote users of the CallManager phone system.

I've looked on the Advanced tools and under the Integration tools of Tools Depots, vut can't see an option that would allow this.

Can it be done?, If so, how?

Thanks,

Chris

6 Replies 6

CHRIS HABBERJAM
Level 1
Level 1

The reason is this.

G729 default sample rate is 20ms, but the customer wants to set it to 30ms for greater efficiency of ATM network !

We don't support fiddling with the sample rates on the supported codecs - you can, of course, change the default recording coded with the SetRecordFormat tool but we don't support changing the sample rate. Your efficiency may be better but your sound quality will take a pretty serious hit by dropping your sample rate by 30% like that. G729a is already around 1KB/sec which is very efficient (certainly better than the 8KB/sec you get with G711).

Are they experiencing problems currently with bandwidth using the default (tested and supported) G729a codec settings? I'd definitely suggest trying it as it is first before fiddling - they will almost certainly be unhappy with the quality if they monkey with it.

Is there a way to select sample rate for 711? I see in several documents that there is a 10ms and a 20ms sample rate, but how does one toggle between them?

Are you referring to the sample rate for the incoming/outgoing audio stream? If so, the feature you are looking for has been supported since Unity 4.0(3):

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/prod_release_note09186a00801b92bc.html#262913

There is no change that needs to be made on Unity, since CallManager sets up the call, it simply tells Unity which packetization interval to use, and it should just work. You just need to make the change in CallManager Service Parameters to configure which packetization interval you want to use for G711 and G729.

However, please note that there is a defect open against Unity 4.0(3) and Unity 4.0(4) that impacts audio quality when using larger packetization intervals. This problem will be addressed in Unity 4.0(5).

http://wwwin-metrics.cisco.com/protected-cgi-bin/ddtsdisp.cgi?id=CSCed52913

Thanks for the response, I've already sent this on to my colleague.

Apologies for taking so long to get back to you, but I haven't heard back from him yet. This looks like it should sort out his problem.

Many thanks

I think I figured it out, correct me if I am wrong.

#codec codec [bytes payload_size]

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