asterisk users - Jan 2005 - What is acceptable network latency for voipconnection?

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 enemy. Jitter is the variation in
packet timing, for example, packet A arrives in 80ms, packet B in 120ms,
and packet C in 70ms. The jitter for this scenario would be 120ms-70ms 50ms. Of
course the jitter time is only half of the story, the number of
packets that are "outliers" in the RTP stream will also have an
impact.
Typical jitter measurements are stated as "average jitter" which helps
masks the problem, if you have 100,000 consistent packets in a row, the
10 slow packets in a row, then back to consistent, the 10 packets are
only .1% of the total but will be heard in the voice stream as a dropout
(the exact number of slow or dropped packets the can be tolerated in a
row is determined by the RTP settings and the devices packet buffers).

There are only two ways to get acceptable performance;

1. use a private or managed link between your VoIP endpoints and
prioritize the RTP streams between the endpoints, leaving the jitter,
delay, and packet loss for the data apps.
Or
2. use public unmanaged links that are way undersubscribed so there is
never any contention for bandwidth, because contention for bandwidth is
he number one cause of jitter, delay, and packet loss.

Most consumer broadband systems do not fall into the undersubscribed
category whereas most T1 and above commercial services are much closer
to undersubscribed. I have seen cable systems and DSL networks that are
oversubscribed at more than 100:1. (too much...).

So the short answer to the question, 100ms or less is desired, but
useless if accompanied by packet loss and jitter.

There are programs you can use to analyze delay, jitter and packet loss.
Search the web for a free one, tune the packet size and rate to match N
(number of active alls) times your RTP parameters to get a better
analysis. Run this for several minutes during a peak period on your
network (7 to 9am and 7 to 10pm for consumer broadband systems).

The result you get is meaningful for that moment, and is no indication
that you will continue to get the same performance.

The real problem comes when cable operators and DSL providers decide to
prioritize RTP for their VoIP customers over VoIP traffic bound for
other providers when the oversubscribed links are congested.


> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of David Liu
> Sent: Saturday, January 08, 2005 8:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] What is acceptable network latency for
> voipconnection?
> 
> Well there is nothing much you can do if you don't own all the routes.
> But in
> concept you can, and this is purely just theoritical and a very
unhealthy
> thing for the Internet, is to write a program running on your router
that
> constantly streams traffic to your end point, this will maintain a
> constant
> bandwidth from your network to your far-end.  Then, your program
should
> detect
> within a few ms that you are setting a call up and immediately reduce
your
> bogus traffic and make room for your "Real" voice traffic.  Again
this
is
> super unhealthy for the Internet, but the idea is TDM on STDM -
constantly
> occupying certain trunks (bandwidth) on the Internet.  So whenever you
> need
> it, you will have it.
> 
> David
> 
> 
> 
> On Sat, 8 Jan 2005 06:22:58 -0800 (PST), Robert Augustyn wrote
> > Very good point.
> > So what can you do ( if anything ) to control the load
> > on the network outside of your control?
> > robert
> >
> > --- David Liu <david@deltapath.com> wrote:
> >
> > > Assuming the network loading is fairly constant,
> > > 300ms latency is actually not
> > > noticeable unless you put both phones next to your
> > > ears to compare.
> > >
> > > Latency affects delay while network loading affects
> > > voice quality (e.g. break
> > > ups) If the either end of your network is
> > > experiencing very bursty traffic
> > > patterns, then even a small latency won't
> > > necessarily guarrantee good sound
> > > quality.
> > >
> > > David Liu
> > > Hong Kong
> > >
> 
> _______________________________________________
> 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

Damon,
Thanks for your comments.
This seems like serious problem, how is it done then
in real world? I cannot see any business moving to
voip when you cannot control quality of service.
Can you recomend any free programs which alow you
analyze delay,jitter and packet loss?
Once again thank you.
robert


--- Damon Estep <damon@suburbanbroadband.net> wrote:
> 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 enemy. Jitter
> is the variation in
> packet timing, for example, packet A arrives in
> 80ms, packet B in 120ms,
> and packet C in 70ms. The jitter for this scenario
> would be 120ms-70ms > 50ms. Of course the jitter time is only half of
the
> story, the number of
> packets that are "outliers" in the RTP stream will
> also have an impact.
> Typical jitter measurements are stated as "average
> jitter" which helps
> masks the problem, if you have 100,000 consistent
> packets in a row, the
> 10 slow packets in a row, then back to consistent,
> the 10 packets are
> only .1% of the total but will be heard in the voice
> stream as a dropout
> (the exact number of slow or dropped packets the can
> be tolerated in a
> row is determined by the RTP settings and the
> devices packet buffers).
> 
> There are only two ways to get acceptable
> performance;
> 
> 1. use a private or managed link between your VoIP
> endpoints and
> prioritize the RTP streams between the endpoints,
> leaving the jitter,
> delay, and packet loss for the data apps.
> Or
> 2. use public unmanaged links that are way
> undersubscribed so there is
> never any contention for bandwidth, because
> contention for bandwidth is
> he number one cause of jitter, delay, and packet
> loss.
> 
> Most consumer broadband systems do not fall into the
> undersubscribed
> category whereas most T1 and above commercial
> services are much closer
> to undersubscribed. I have seen cable systems and
> DSL networks that are
> oversubscribed at more than 100:1. (too much...).
> 
> So the short answer to the question, 100ms or less
> is desired, but
> useless if accompanied by packet loss and jitter.
> 
> There are programs you can use to analyze delay,
> jitter and packet loss.
> Search the web for a free one, tune the packet size
> and rate to match N
> (number of active alls) times your RTP parameters to
> get a better
> analysis. Run this for several minutes during a peak
> period on your
> network (7 to 9am and 7 to 10pm for consumer
> broadband systems).
> 
> The result you get is meaningful for that moment,
> and is no indication
> that you will continue to get the same performance.
> 
> The real problem comes when cable operators and DSL
> providers decide to
> prioritize RTP for their VoIP customers over VoIP
> traffic bound for
> other providers when the oversubscribed links are
> congested.
> 
> 
> 
> > -----Original Message-----
> > From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-
> > bounces@lists.digium.com] On Behalf Of David Liu
> > Sent: Saturday, January 08, 2005 8:01 AM
> > To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> > Subject: Re: [Asterisk-Users] What is acceptable
> network latency for
> > voipconnection?
> > 
> > Well there is nothing much you can do if you don't
> own all the routes.
> > But in
> > concept you can, and this is purely just
> theoritical and a very
> unhealthy
> > thing for the Internet, is to write a program
> running on your router
> that
> > constantly streams traffic to your end point, this
> will maintain a
> > constant
> > bandwidth from your network to your far-end. 
> Then, your program
> should
> > detect
> > within a few ms that you are setting a call up and
> immediately reduce
> your
> > bogus traffic and make room for your "Real" voice
> traffic.  Again this
> is
> > super unhealthy for the Internet, but the idea is
> TDM on STDM -
> constantly
> > occupying certain trunks (bandwidth) on the
> Internet.  So whenever you
> > need
> > it, you will have it.
> > 
> > David
> > 
> > 
> > 
> > On Sat, 8 Jan 2005 06:22:58 -0800 (PST), Robert
> Augustyn wrote
> > > Very good point.
> > > So what can you do ( if anything ) to control
> the load
> > > on the network outside of your control?
> > > robert
> > >
> > > --- David Liu <david@deltapath.com> wrote:
> > >
> > > > Assuming the network loading is fairly
> constant,
> > > > 300ms latency is actually not
> > > > noticeable unless you put both phones next to
> your
> > > > ears to compare.
> > > >
> > > > Latency affects delay while network loading
> affects
> > > > voice quality (e.g. break
> > > > ups) If the either end of your network is
> > > > experiencing very bursty traffic
> > > > patterns, then even a small latency won't
> > > > necessarily guarrantee good sound
> > > > quality.
> > > >
> > > > David Liu
> > > > Hong Kong
> > > >
> > 
> > _______________________________________________
> > 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
> _______________________________________________
> 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
>