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Unread 01-17-2009, 11:52 AM   #2
carlosalbffgomes
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SIP, when used in VoIP applications, is mostly a P2P protocol, so usually audio goes directly from source to destination, not through SIP proxy, which is used only to initiate the multimedia session. In this scenario, if you are making an overseas call, and the SIP server you are using is on the other side of the globe, your audio will not suffer from high latency because the two calling points are not so very far away from each other. The problem is when there is a need to use an audio proxy for transcoding or solving NAT issues. In this last scenario you should consider the latency introduced by the proxy being used. Some providers acts as audio proxying services, some don't. There is no universal rule, so you should understand the problem and have the possibility to know which are the technical specifications of the service being used. However, this is an information that most commercial departments can't give you, so you should discover it for your self through some testing of the provider.
Regards.
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