How does an ISP provide a Jitter SLA on a Data T1? Jitter < 5ms? How does one measure that?
We're an ISP, we've been doing T1's for many years. I know that customers can ping any equipment or servers within our network with < 10ms response times, we link with 3 large Tier 1 providers, DS3 speeds for 2 of them and a 90mbs NMLI for the third, we're not over-saturated at all, bandwidth to spare, and our customers generally report 50ms average response times out to the internet. To further regions and when going through a couple of networks, it can be up to 80ms. The only time I've seen over 100ms is to international destinations, but the ping response times generally stay consistent, no dramatic spikes. That's with a standard frame relay T1. Point to Point T1's are slightly better. However I didn't create or design nor do I maintain this company's network, I just do the VOIP thing, so I'm quite curious to see their SLA, I wonder if a misconfiguration somewhere could affect quality and give me a head-ache. For a few of our heavy VOIP customers, we use 2 Point-To-Point T1's from our location to theirs, we then combine traffic to go across both T1's, if 1 T1 fails, it'd all automatically go across the remaining T1. They can ping our Asterisk server with an average of 10ms latency. We also tend to use private IP's, to avoid internet DDOS's and worms and such. We've been able to push 35-40 calls across this link. Probably more if I used trunking and/or switched codecs from ULaw. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Augustyn Sent: Monday, January 10, 2005 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] What is acceptable networklatencyforvoipconnection? 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 [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
