Voice Traffic Requirements

control

Have read the following -

Voice traffic has the following additional one-way requirements........
End to end delay: 150ms or less
Jitter: 30ms or less
Packet Loss: 1% or less

The bit i don't understand is the End to End delay. We have sites that that the delay / round trip time is 600ms + (VSAT) but voice between Head Office and these site calls work ok...?

Is this just a "nice to have" end to end delay that is stated?

pitviper

150ms is the ITU standard. Depending on what you read, Cisco states that one way delays up to 200ms are perfectly acceptable. Sure, you can get away with much more without quality degradation, but you certainly wouldn’t want to design your network under the assumption that you *should* be OK.

control

Cheers PitViper..

Another q if you may..

The end to end delay, is this purely between phone to phone, or call manager to call manager? For example, the 600+ sites that I am talking about, actually register their against the head office call manager. So although the sites are remote to each other, the phones are both registered to head office.

Although I'm guessing this doesn't come into the actual conversation, as the Call Manager drops out the equation once the phones are connected and use RTP? That correct?

pitviper

The delay is for phone to phone (or site to site is close enough). The CUCM instructs the phones on what to do each step of the way, then the RTP stream is created between phones.