if point to poin, try to do a large file transfer (multiple connections) between the two sites if possible. check the average bandwidth that you can get plus packet loss. compare the average bandwidth with nominal ethernet bandwidth (based on audio and video codec used) x num of desired simultaneous call.
On 10/19/07, Marvin T. Pascual <[EMAIL PROTECTED]> wrote: > > Hello all, > > How will you determine the voice (and video) quality of SIP calls on two > PBX peers or from your PBX to the ITSP? Automated latency tests is one. > Any others? > > Considering LCR is the preferred way but if the voice (and video) quality > is poor, I would prefer calling through PSTN trunk(s) and pay additional > cost for the long distance calls. > > Regards, > > Marvin > > _________________________________________________ > Philippine Linux Users' Group (PLUG) Mailing List > [email protected] (#PLUG @ irc.free.net.ph) > Read the Guidelines: http://linux.org.ph/lists > Searchable Archives: http://archives.free.net.ph >
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