Bandwidth Used by a single Call

dconstantino · ‎10-15-2007

I have been doing IP Tel for a long Time and I have always read and been told G729 is around 24 to 32K per call.

But after working with IP to IP gateways and SIP and some very large enterprise customers and talking to TAC this is refering to a single RTP stream from point A to point B then as the converstion changes from point B to A that is a RTP Stream which really makes that one call 64k.

I found out that the Bandwidth Calculator calculates based on a single oneway RTP stream. So when you tell a customer a g729 call is 24 to 32k is not correct it is more like two 32k converstions that in reality make up 64k.

So a 512K CAR will only handle 8 calls at 32k per stream. 512K/64

I hope you are not as confused not as I have become. And why can't the BW Calculators just do the total BW for 8 calls.

bgibson · ‎10-15-2007

That isn't exactly true.

Duplex RTP streams will be generally the same size in both directions. Whatever compression you use the stream will be the same size in both directions. The only variation will be based on the amount he side actually inputs into the call.

However that doesn't really impact your bandwidth design since QoS policies only deal with one direction, usually outbound. If your median call utilization is 32Kb/s and you have a 512 Kb/s queue for voice then you probably can get about 14 calls without breaking a sweat. If your QoS is setup for one way calls in both directions then you will be fine.

Now of course the game changes a bit if you are dealing with asymmetric pipes such as DSL but that is more a planning issue than anything else.

dconstantino · ‎10-16-2007

I thought this way also but the way I am handling voice is with a PQ and a Gold CAR from the ISP of 512K on a 1.5m circuit.

I Call consist of 2 call legs from the IP Phone to the Gateway Leg 1 and from the gateway to the PSTN Leg 2. That is a 32k call.

Then the stream back from the PSTN to the Gateway Leg 3 and then to the phone Leg 4is another call 32K. So a conversation is 64k.

bgibson · ‎10-16-2007

So let me see if I understand.

You have something like this...

Phone -> WAN -> Gateway -> WAN -> SIP Gateway?

I can see how that would be a problem for you.

I'm not really sure why you have it set up that way though.

Why not set it up

Phone -> Gateway -> WAN -> SIP Gateway?

dconstantino · ‎10-16-2007

Here is the RTP Stream path.

Phone--->MPLS--->IPIPGW--->MPLS--->ISPSIPGW

--->PSTN---->

I think were I am confused is that th gateway has to setup an RTP stream from the phone to the gateway then the conversation coming back setup another RTP stream inbound. so 32k+32K=64k