Hi All, How can I find out One way latency from my PBX to my SIP Trunk Provider. My SIP provider recommends a One way latency of 100ms for good Voice quality. Ping request to their IP Address gives me a response in approx. 260ms. Will that be good enough for a SIP Trunk. Please help. We are trying to sign up with a new SIP Provider. Thanks, Najim -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111201/1820cf4f/attachment.htm>
Am 30.11.2011 21:47, schrieb NaJIm:> Hi All, > > How can I find out One way latency from my PBX to my SIP Trunk Provider. > My SIP provider recommends a One way latency of 100ms for good Voice > quality. Ping request to their IP Address gives me a response in approx. > 260ms. > Will that be good enough for a SIP Trunk. > > Please help. We are trying to sign up with a new SIP Provider. > > Thanks, > Najim > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersHi Najim, a ping is the time a packet needs for travelling to a destination and back to you. So the one way latency you are refering to, should be half the time your ping took. In your case this will be 130ms, I would say this is still reasonable. regards, Ruben
Thank you Ruben. Is there anything else that I should be concerned about when looking for a SIP provider. ?? Regards, Najim. On Thu, Dec 1, 2011 at 2:34 AM, Ruben R?gels <ruben.roegels at jumping-frog.org> wrote:> Am 30.11.2011 21:47, schrieb NaJIm: > > Hi All, > > > > How can I find out One way latency from my PBX to my SIP Trunk Provider. > > My SIP provider recommends a One way latency of 100ms for good Voice > > quality. Ping request to their IP Address gives me a response in approx. > > 260ms. > > Will that be good enough for a SIP Trunk. > > > > Please help. We are trying to sign up with a new SIP Provider. > > > > Thanks, > > Najim > > > > > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > Hi Najim, > > a ping is the time a packet needs for travelling to a destination and > back to you. So the one way latency you are refering to, should be half > the time your ping took. > > In your case this will be 130ms, I would say this is still reasonable. > > > regards, > Ruben > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111201/5c007ad5/attachment.htm>
My ping requests show 0% packet loss. How do we find out packet re-ordering.?? Najim. On Thu, Dec 1, 2011 at 5:18 AM, Hans Witvliet <asterisk at a-domani.nl> wrote:> On Thu, 2011-12-01 at 04:52 +0530, NaJIm wrote: > > Is there anything else that I should be concerned about, when looking > > to signup for a SIP provider. ?? > Latency is important, but packet loss also, likewise packet re-ordering. > > hw > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111201/68dc8606/attachment.htm>