similar to: What is acceptable network latency forvoipconnection?

That "program" will be detected by your ISP within a day or so, determined to be a virus, and your service will get disconnected...which n turn will not help your latency or jitter at all. VoIP can tolerate a fair amount of latency; latency over about 100ms is heard as a perceptible delay resulting in a connection that appears to be half duplex. Jitter, on the other had, is the real

There is a theory that says that jitter buffers should not be used until the end of the voice path where jitter might be introduced. With that in mind, and in this scenario, the jitter buffers should reside at the ATA and media gateway; ATA (SIP UA) <> ASTERISK NATIVE BRIDGE <> MEDIA GATEWAY (SIP TO TDM) That raises a question about the Asterisk Native Bridge; Are the UDP RTP

Ankur Arora <ankur.a.arora at oracle.com> writes: > A KVM host (or another hypervisor) might advertise paravirtualized > features and optimization hints (ex KVM_HINTS_REALTIME) which might > become stale over the lifetime of the guest. For instance, the > host might go from being undersubscribed to being oversubscribed > (or the other way round) and it would make sense for the

> How does an ISP provide a Jitter SLA on a Data T1? Jitter < 5ms? How does > one measure that? You can get a good feel for delay and jitter just by running a continuous ping to a core router on your ISPs network during peak times(or to Google for that matter) and visually monitoring the results. A good, unsaturated link will have extremely consistent response times with less than 5ms

http://www.asterisk.org/node/48317 does a nice job of explaining the 1.4 jitter buffer, however it raised a question in my mind. In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the UDP RTP packets renumbered on transmit, or is the original sequence number preserved in the UDP header? A comment is made on the referenced blog that jitter buffering is best implemented at the

Juan, I am running a Calling Card application on a Dell PowerEdge 2850 with Asterisk 1.0.7. Recording conversations I have seen on my server causes the processors to burn more than necessary so I would recommend what William from Signate recommended: " Consider saving recorded calls in a database on a separate server. It will be simpler to build a retrieval interface that does not

Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should

On Tue, Apr 07, 2020 at 10:02:57PM -0700, Ankur Arora wrote: > A KVM host (or another hypervisor) might advertise paravirtualized > features and optimization hints (ex KVM_HINTS_REALTIME) which might > become stale over the lifetime of the guest. For instance, the > host might go from being undersubscribed to being oversubscribed > (or the other way round) and it would make sense

On Tue, Apr 07, 2020 at 10:02:57PM -0700, Ankur Arora wrote: > A KVM host (or another hypervisor) might advertise paravirtualized > features and optimization hints (ex KVM_HINTS_REALTIME) which might > become stale over the lifetime of the guest. For instance, the > host might go from being undersubscribed to being oversubscribed > (or the other way round) and it would make sense

If SLA supports IP trunks, can shared extension appearance be achieved using a local SIP trunk in place of an extension? Basically, I'm trying to allow some stations (Polycom IP 650) to have a shared extension amongst all of them. Ideally, I'd like for the LED to show if that extension is in use, and I'd like for the extension to ring all stations on that extension when a call comes

I configured my sla.conf to use with a Polycom phone. I have no idea if I did it right, however, none of the console "sla" commands exist. Do I have to something special to compile in this support, or should it just work out of the box? ~jay

Il 02/08/20 18:54, Stephen John Smoogen ha scritto: > On a side note, you keep emphasizing you aren't expecting an SLA.. but > all your questions are what someone asks to have in a defined SLA. I > have done the same thing in the past when things have gone badly, but > couching it in 'I am not asking' just makes the people being asked > grumpy. Better to be open and say

Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a "SAY NUMBER 123 #" before and after

I would be incredibly grateful to anyone who'll help me translate some SAS code into R code. Say for example that I have a dataset named "dat1" that includes five variables: wshed, site, species, bda, and sla. I can calculate with the following SAS code the mean, CV, se, and number of observations of "bda" and "sla" for each combination of

Greetings, Back in September someone asked about documentation for the new SLA feature in 1.4, however they received no replies. I thought I might ask the same question now in December. Apart from sla.conf.sample and a few comments in app_meetme.c I have been unable to find useful documentation. Is anyone using this feature right now? Is there a helpful source for information this highly

Hello, I am looking for a "real life" example of using SLA lines under Asterisk. I'll describe my environment and would like to know how I define it in Asterisk (version 1.4 final). Suppose I have two multi lines phones. The first phone has extension 1 assigned to it, and the second phone has extension 2 assigned to it. Now, I want extension 3 to be available on both phones as

I understand that the Asterisk SLA implementation is somewhat different from most key systems and PBX systems. I also understand that in Asterisk, one does not put an SLA line on hold since it is just a MeetMe conference. However, is there any way to make the BLF flash when the answering party on the Asterisk system presses the hold key on their set and leaves the calling party alone in the

anyone using/experimenting with this new feature in asterisk 1.4? is anybody able to post some info how to use and what features are supported? I have general knowledge how SLA should work, ie. monitor status of another line like BLF with additional features like answer ringing call, barge into existing call on shared line and make conference call or resume call, that was put on hold on

Has anyone done SLA with Polycom phones? I've got a large project coming up where the customer is keen on SLA for trunks and extensions. Trunks will be on a PRI. We may do this with Cisco phones if they work better. Mark Willis

Size: - one location - 15 IP Phones ( 1 dect) - Create new voip trunk (current are ISDN) (30 number block) - LTS is important - an SLA is optional at the moment there is no one On 11.12.2017 22:31, Ron Wheeler wrote: > You might want to add some details > - size of the project > ?-- number of locations > ?-- number of extensions > - are you converting your trunks? > - what are