Can I transcode a g.729 call to use a T1 channel.

[Michael Ricci]

As I am sure you are all aware, T1 channel's up 64kbps of bandwidth. I have a few users who are set up to use the G.729 codec vs. the G.711 codec everyone else uses. When they attempt to place a call to the PSTN, the call goes through but no audio is heard on either end.

I believe this is because they are trying to utlilize the T1 using the 24kbpb that G.729 converts the stream and overhead to, is there anyway I can transcode the audio stream back to 64kbps before it attempts to cross the T1? I think this will fix the problem.

[dijohn]

Is this your call flow?

IP Phone ----- Voice GW --- PRI -- PSTN?

In this case, I dont think the T1/E1 has anything to do with your oneway/no way audio..

When you make a call from IP Network to the PSTN out the PRI, DSPs are used to xmit the same out the PRI.

g711u/g711a = 15 MIPS per call
g729a/g729ab = 30 MIPS per call
g729/g729b = 40 MIPS per call

So if you have a phone talking G711, we would use 15 MIPS from the DSP. PVDM2-16(15 x 16 G711 calls)  has 240 MIPS.

Do we have a routin issue?.. Do one thing...

Make a test call, collect

show voip rtp connection

show call active voice brief

debug isdn q931

debug mgcp packets ( if its a mgcp gw)

debug ccsip messages ( if its a sip gw)

debug h225 asn1(h323)

debug h245 asn1 (h323)

/divin

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