RE: [Asterisk-Users] What is acceptable network latencyforvoipconnection?
Damon, Thanks that is great info. robert --- Damon Estep [EMAIL PROTECTED] wrote: Thanks, So what are the fresholds of the jitter, delay, and packet loss I should be asking my ISP for? robert On a T1 you should expect an SLA that states; Latency - 80ms round trip latency between your end and the ISP core routers within a few thousand miles, SLAs typically only cover traffic on the ISPS network. Try to use the same ISP for all VoIP endpoints so there is never any question about which network the delay occurs on. Jitter - SLA should be 5ms Packet loss - should be 0.5% While VoIP might tolerate more jitter and packet loss than stated above, these are not unreasonable parameters for T1 service and you should be able to get an SLA that falls in or near these parameters. Any ISP that can not provide a close SLA is not confident in their network. In my experience, packet loss is the biggest enemy, jitter is a close second, and latency is third (if it is still under ~200ms). Latency is a function of bandwidth and without congestion will be very consistent on similar bandwidth connections. Increases in latency due to congestion almost always come with high packet loss and jitter. Google for T1 Internet SLA and you will see many sample SLAs. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What is acceptable network latencyforvoipconnection?
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. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What is acceptable network latencyforvoipconnection?
Thanks, So what are the fresholds of the jitter, delay, and packet loss I should be asking my ISP for? robert On a T1 you should expect an SLA that states; Latency - 80ms round trip latency between your end and the ISP core routers within a few thousand miles, SLAs typically only cover traffic on the ISPS network. Try to use the same ISP for all VoIP endpoints so there is never any question about which network the delay occurs on. Jitter - SLA should be 5ms Packet loss - should be 0.5% While VoIP might tolerate more jitter and packet loss than stated above, these are not unreasonable parameters for T1 service and you should be able to get an SLA that falls in or near these parameters. Any ISP that can not provide a close SLA is not confident in their network. In my experience, packet loss is the biggest enemy, jitter is a close second, and latency is third (if it is still under ~200ms). Latency is a function of bandwidth and without congestion will be very consistent on similar bandwidth connections. Increases in latency due to congestion almost always come with high packet loss and jitter. Google for T1 Internet SLA and you will see many sample SLAs. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
