Hi,
These are my points. Any comments welcome.
For good voice quality the round trip delay in the voice conversation should be below 400 ms. Any thing above 500 ms delay is not at all acceptable. A normal voice conversation needs around 87 kbps bandwidth.
Now think about how a normal VoIP system works. Between USA and INDIA, there will be a round trip delay of around 350 ms for Internet. Normally VoIP IP phones will either utilize SIP or H323 based systems. SIP or H323 will add an additional delay of 75 ms on each end. Additionally H323 and SIP will utilize some compression technique to save bandwidth from 87 kbps to 32 kbps with G711 or to 20 kbps with G723. So this compression and decompression will add another 75 ms delay in each end.
Now the total delay comes to around 650 ms, which makes the quality of the VoIP IP phone unacceptable. Also the central signaling server should handle all the voice traffic. Therefor during high call volume this delay will increase. In addition to affecting delay compression and decompression degrade the voice quality.
So the for an effective VOIP callcente setup, we have to avoid these delays. A better VOIP system will be the one which is not using the gateways and compression techniques. The IP PBX should be only used for finding the IP address of the IP phone and after that the communication between IP phone and the US POTS port should be direct.
Thanks,
Vasu
www.datasoftnet.com
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