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Live sip calls take up bandwidth ! as our voices travel both directions.
does dialing take up bandwidth too ? isn't it supposed to be just signaling ?
if it takes bandwidth ... how much does it take 1% compared to size taken by live call ? 5% ?
voice is compressed in voip ... is signaling also compressed in voip ?

I rise this topic because its crucial decision to know how much bandwidth one needs for his calls, does one need to take into account 1.5 the number of his agents when calculating bandwidth or only the number of agents + (only) the bandwidth taken by dailing process (signaling).

hope you guys share your experiences because we buy the bandwidth that goes with our number of lines that goes with our number of agents .... but do we really have to ?

Final question, what is your experience with simultaneouse calls ? if you have 100 agent & you follow lines usage you will find out you don't go over 60% of the number of lines you bought. is it wiser to buy less lines ?, I mean take into account the number of simultaneouse calls instead of the number of agents ?

hope every one gives us his opinion.

I don't deal that much with this problem since most of my large clients use T1 lines or dedicated DS3 circuits here in the US, but I will say that whenever you try to get "just enough" bandwidth to meet your needs, it will not be enough.

And most places where bandwidth is scarce or in poor quality you will run into latency and throughput consistency issues and you will want to have as much bandwidth as you can get to guarantee good audio quality.

Matt, I am talking about International dedicated links which cost a bit because they are guaranteed, add to that dedicated equipment that compress voice to 8K. which means if you buy 512k you get 64 simultaneouse calls (512/8=64). the next option is to buy 1024K, you can't buy chunks of 8K.
so understanding the exact bandwidth taken by the call & the dailing is very important since if dialing doesn't take bandwidth then no need to pay for bandwidth not used.
what is your % of line occupation (simultaneouse calls) ? in my case we hardly bypass 60% of total agents.

I have not really done any studies of this, it would be interesting to see more information on this if you end up doing more analysis. I would think that the quality of the leadbase would greatly affect the dialed lines in use ratio.

This issue is a common subject when speaking to clients about the differences of connectivity. Unlike a PRI which doesn't utilize a channel unless there is a connected call, VOIP requires it's allocated bandwidth to send and receive information. Here is a breakdown of the G.729 compressed codec.

G.729 call with voice payload size of 20 bytes (20 ms): (40 bytes of IP/UDP/RTP headers + 20
bytes voice payload)* 8 bits per byte * 50 pps = 24 Kbps

As you may know, G.711 uses 86Kbps/call.


Matthew Martin
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