[Asterisk-Users] What is acceptable network latencyforvoipconnection?

[Damon Estep]

> > In the real world (or at least in my world) we use undersubscribed
> > internet connections that come with a service level agreement (SLA)
that
> > guarantees that the jitter, delay, and packet loss with be within
> > defined parameters in the service agreement.
> [...]
> 
> In the real world (or imaginary world) that will not include your
> traffic which leaves or enters your ISP as your ISP has no control
> over that aspect.
> 
A good ISP will prioritize RTP on their interface facing you (if you ask
them to) and you can prioritize RTP on your interface facing them. When
doing so it helps to identify the priority traffic by type (RTP) and
destination so other RTP streams do not impact your VoIP traffic. If the
ISP is not too oversubscribed on their upstream link, and you use this
same strategy on your other endpoints, you will get good reliable
results. If you want to check up on your SLA make sure your test tool
traffic is also in the priority queue group.

This is not a new question or answer; we have been doing this for 10
years, h.323 video conferences then, SIP audio and h.323 video now. We
have seen numerous ISPs, and we are also an ISP, all internet
connections are not created equal, my point is; find a good one, and
most good ones provide a good SLA because they know they can meet it.

A "good" ISP alone will fix most RTP issues, and priority queuing will
protect your RTP from your own network.