similar to: What is acceptable network latency for voipconnection?

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. With most DSL and Cable you will not get a SLA, with the cheapest T1s you might get one, but the only penalty to the ISP if they do not meet is a

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

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

Hi! I just wondered if anyone would mine posting their successful jitter buffer settings here for me if they get a moment ?? I've spent a few hours trying to set the jitter buffer up reasonably logically and can definitely tell it makes a difference and can introduce latency and echo if setup incorrectly but I can't see a good post anywhere describing properly what the three settings

Hi Jean-Marc, Thanks for your response. Given a worst case scenario, what is the "worst case" latency (in terms of Speex frames) that the jitter buffer algorithm will incur? We're trying to determine the worst case hard number. Sorry for unclear question below; what I was trying to ask is that given a worst case latency (which I'm asking in the first question) inherent in

Hi, Our project is using the jitter buffer feature built in Speex. We noticed there are some latency when using the jitter buffer. Does anyone know what is the "worst case" latency inherent in the jitter buffer algorithm? I believe someone already mentioned that it's adaptive but is there a worst case hard number (in terms of 20ms Speex frames)? I'm not familiar with the

Hi, I'm trying to use the jitter buffer feature that comes with Speex but I'm getting unexpected latency. I wrote a client application that does VOIP-like functions and without using jitter buffer, the end-to-end latency is around 250 ms (I'm using lowband 5.97 kpbs). However, when I tried to incorporate the jitter buffer feature, the latency would grow as time elapsed (up to a few

We're investigating audio quality issues in our system; maybe someone can help. We're using Asterisk as a basic PBX, with a single PRI on one side and SIP phones on the other: Sipura SPA-841's. We're experiencing several audio effects which seem to commonly correspond to network failures (packet loss, high jitter, etc manifested as "robot voice", dropouts, periodic

[Just curious, and seizing the opportunity to communicate with other folks who are doing the same kind of thing I am...] How are you measuring the latency? I tried measuring it with my program (also Win32-based, also using DirectSound[Capture]) and came up with around 130ms. To measure it, I placed the mic near a speaker to get feedback going, had my program connect to itself (local

Hi, Does anybody know any software that give me Latencty, Jitter and Lost packets to analyze my Call quality ??? Luis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070831/47350d13/attachment.htm

Hi Jean-Marc, Regarding you points: 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is

OK, background/config. running * (show version reports 0.9.0) on Mandrake 9.2 (kernel: 2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card, no IRQ sharing I can find (cat /proc/pci & cat /proc/interrupts), vmstat reports a minimum of 80+% CPU idle when problem occurs. connect to a Grandstream 101 (GS) via vpn (no nat). Link has 100ms - 150ms ROUND TRIP latency

On 08.04.20 07:02, 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 Then this hint is wrong if it can't be guaranteed. > host might go from being undersubscribed to being oversubscribed > (or the other way

In my experience most of the jitter related issues are because people are using too small of audio buffer sizes that match the framing size of Speex - particularly in Windows. This isn't a problem with Speex, but as a programmer you should collect and append a few frames to match the size of your output audio frame buffer before attempting to play the sound. -----Original Message----- From:

> FYI: The below is just my interpretation of the code, I might be wrong. Most of it is right. Actually, would you mind if I use part of your email for documenting the jitter buffer in the manual? > Each time a new packet arrives, the jitter buffer calculates how far ahead > or behind the "current" timestamp it is; this is called arrival_margin. > The "current"

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter

The client is the adobe flash player. No install and on 98% of all desktops but we can't change it. It works ok if people use headphones but we need to stop the howl than can build up if more than one person in a conference has mic to close to speakers. Any ideas? Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > 1) Is it ok if the audio is encoded (using

Hi, I need to have some SIP extentions on remote places where the latency from my asterisk box with public ip is 1~1.5 seconds. What codec will work fine on this sceneary? I'm planning to use iLBC, is a good choice? Regards, Guillermo.

Hello, I understand that only the destination of a call should do jitter buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no transfers), PhoneA and PhoneB need to perform their own jitter buffering, and Asterisk will just forward the frames, correct? What happens if the peer does not support jitter buffering, but is close by so there's no need for jitter buffering? My