asterisk users - Apr 2007 - SIP devices with packet loss tolerance

Greetings list,

Hoping someone might have experience with poorly-performing net connections and
which devices work best over them.

One of our clients has a number of employees that work from home, and are given
a SIP phone to take with them and hook up to their broadband. For the most part,
this works fine, but there are an increasing number where sound quality is poor
("chops" in and out, generally only noticeable to the listener at the
other end, not the employee). Logic suggests it's an upstream bandwidth
issue, so we asked them to try when all other devices were turned off (to cut
out the "kids using bitTorrent" issues), but even with the phone the
only device, call quality was still poor.

Since the connections aren't paid for by the client, we aren't in a
position to mandate particular providers or speeds, but in each case, the
minimum was a 1mb/256k up ADSL. We asked the employees to run some speed tests
to determine real-world speeds, and in each case upstream was around 220-235k (a
little off the "official speed" but not bad). Certainly way more than
the ~35kbps necessary for a g729 call, even with packet overheads.

We've also tested the connections with a constant ping, and latency for
nearly all of them is sub-35ms.

So, that leads me towards packet loss as the only thing left. Generally
speaking, these connections are giving between 1 and 4% packet loss.

Therefore, 3 questions:
1) is this level of packet loss likely to have the effect we're seeing?

2) If so, are there any phones people have tried with particularly good jitter
buffering? If not, any ideas what else might be causing the issue.

3) are some codecs naturally more "tolerant" of jitter than others?
i.e. would there be an advantage to using something apart from g729, and if so,
what would you recommend?

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it/chris.html
This email is made from 100% recycled electrons

Some codecs are more tolerant of packet loss then others, but I don't think
that the type of codec will have a major effect on its ability to deal with
jitter.  Jitter buffers will help but with the side effect of increasing the
overall latency of the conversation (hence the buffer).  Lost packets have
the largest effect on codecs which transmit with a high audio length to
packet ratio.  Since g729 transmits only 10ms of audio per packet, I would
expect lost packets to have less of an impact then they would on, say, and
iLBC conversation where 30ms of audio is placed in each packet.  The length
of the audio pay load may also effect the symptoms of jitter, but I can't
really speak to that more than anecdotally.

g729 is one of the more expensive codecs for audio conversion purposes.
Have you taken a look at your server load when poor quality was reported?



On 4/23/07, Chris Bagnall <lists@minotaur.cc>
wrote:
>
> Greetings list,
>
> Hoping someone might have experience with poorly-performing net
> connections and which devices work best over them.
>
> One of our clients has a number of employees that work from home, and are
> given a SIP phone to take with them and hook up to their broadband. For the
> most part, this works fine, but there are an increasing number where sound
> quality is poor ("chops" in and out, generally only noticeable to
the
> listener at the other end, not the employee). Logic suggests it's an
> upstream bandwidth issue, so we asked them to try when all other devices
> were turned off (to cut out the "kids using bitTorrent" issues),
but even
> with the phone the only device, call quality was still poor.
>
> Since the connections aren't paid for by the client, we aren't in a
> position to mandate particular providers or speeds, but in each case, the
> minimum was a 1mb/256k up ADSL. We asked the employees to run some speed
> tests to determine real-world speeds, and in each case upstream was around
> 220-235k (a little off the "official speed" but not bad).
Certainly way more
> than the ~35kbps necessary for a g729 call, even with packet overheads.
>
> We've also tested the connections with a constant ping, and latency for
> nearly all of them is sub-35ms.
>
> So, that leads me towards packet loss as the only thing left. Generally
> speaking, these connections are giving between 1 and 4% packet loss.
>
> Therefore, 3 questions:
> 1) is this level of packet loss likely to have the effect we're seeing?
>
> 2) If so, are there any phones people have tried with particularly good
> jitter buffering? If not, any ideas what else might be causing the issue.
>
> 3) are some codecs naturally more "tolerant" of jitter than
others? i.e.
> would there be an advantage to using something apart from g729, and if so,
> what would you recommend?
>
> Regards,
>
> Chris
> --
> C.M. Bagnall, Director, Minotaur I.T. Limited
> For full contact details visit http://www.minotaur.it/chris.html
> This email is made from 100% recycled electrons
>
>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
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> To UNSUBSCRIBE or update options visit:
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On Mon, 23 Apr 2007, Chris Bagnall wrote:
> Greetings list,
>
> Hoping someone might have experience with poorly-performing net 
> connections and which devices work best over them.
>
> One of our clients has a number of employees that work from home, and 
> are given a SIP phone to take with them and hook up to their broadband. 
> For the most part, this works fine, but there are an increasing number 
> where sound quality is poor ("chops" in and out, generally only 
> noticeable to the listener at the other end, not the employee). Logic 
> suggests it's an upstream bandwidth issue, so we asked them to try when
> all other devices were turned off (to cut out the "kids using 
> bitTorrent" issues), but even with the phone the only device, call 
> quality was still poor.
>
> Since the connections aren't paid for by the client, we aren't in a
> position to mandate particular providers or speeds, but in each case, 
> the minimum was a 1mb/256k up ADSL. We asked the employees to run some 
> speed tests to determine real-world speeds, and in each case upstream 
> was around 220-235k (a little off the "official speed" but not
bad).
> Certainly way more than the ~35kbps necessary for a g729 call, even with 
> packet overheads.
>
> We've also tested the connections with a constant ping, and latency for
> nearly all of them is sub-35ms.
>
> So, that leads me towards packet loss as the only thing left. Generally 
> speaking, these connections are giving between 1 and 4% packet loss.
For (what I'm assuming is a UK ADSL connection), that packet loss is very 
high. Is it loss to their head-office where the SIP server is, or are they 
using some external hosted SIP service?

I don't see any packet loss from my home ADSL line, so something is 
"fishy"...
> Therefore, 3 questions: 1) is this level of packet loss likely to have 
> the effect we're seeing?
Generally speaking with one packet every 20ms, 1% loss is a dropped packet 
every 2 seconds. You'd barely notice it unless it was regular. 4% loss is 
a packet every 0.5 seconds. 1% would be an annoying click every now & 
then, 4% will sound a bit ropey.
> 2) If so, are there any phones people have tried with particularly good 
> jitter buffering? If not, any ideas what else might be causing the 
> issue.
>
> 3) are some codecs naturally more "tolerant" of jitter than
others? i.e.
> would there be an advantage to using something apart from g729, and if 
> so, what would you recommend?
Changing ISP.

Maybe not an acceptable solution, but on a quiet line, I'd find it hard to 
justify a constant 1-4% packet loss, however I could belive that a dodgy 
el-cheapo ISP for the masses would have issues - espeically with high 
levels of small packets....

You might also want to check the router at head-office, if it's an 
in-house hosted service. Make sure they have a good router that can handle 
the increased packet load..

Gordon

Hi:

Chris Bagnall wrote:
> One of our clients has a number of employees that work from home, and
> are given a SIP phone to take with them and hook up to their
> broadband. For the most part, this works fine, but there are an
> increasing number where sound quality is poor ("chops" in and
out,
> generally only noticeable to the listener at the other end, not the
> employee). Logic suggests it's an upstream bandwidth issue, so we
> asked them to try when all other devices were turned off (to cut out
> the "kids using bitTorrent" issues), but even with the phone the
only
> device, call quality was still poor.
> 
> Since the connections aren't paid for by the client, we aren't in a
> position to mandate particular providers or speeds, but in each case,
> the minimum was a 1mb/256k up ADSL. We asked the employees to run
> some speed tests to determine real-world speeds, and in each case
> upstream was around 220-235k (a little off the "official speed"
but
> not bad). Certainly way more than the ~35kbps necessary for a g729
> call, even with packet overheads.
I have long been suspicious of bandwidth statistics. In my experience,
just because a provider says the bandwidth is 256kbps doesn't mean it
is, by a long way. As a general guideline, upstream bandwidth on
residential broadband connections is poorly served. That's the dirty
secret; as more people are enticed to go to VoIP for residential phone
service, that limitation is going to become painfully apparent.

How did you perform the speed tests? The freely available ones on the
public Internet leave a lot to be desired. If you want to get truly
reliable performance numbers, use iperf. There's a Win32 version available:

http://dast.nlanr.net/Projects/Iperf/

You can put this on your Asterisk server and do some really meaningful
throughput tests to and from your roaming users. At the research network
operator where I used to work, we used this all the time for
troubleshooting. Even though we had an extremely expensive network that
was fibreoptic over its entire length, performance problems were common
and the thing required constant coddling. (As a point of comparison, the
best I've seen out of a copper gigabit Ethernet (in a laboratory
environment, no less) is 330 Mbps -- that's a long way from gigabit
speeds, and I seriously doubt we're talking about 600 Mbps of network
overhead.)

On the matter of the BitTorrent factor: did you have the users connect
the phone, and only the phone, to the Internet connection?
> We've also tested the connections with a constant ping, and latency
> for nearly all of them is sub-35ms.
The connections are simply not optimized for real-time performance. I
don't know that there's much you can do about this.
> So, that leads me towards packet loss as the only thing left.
> Generally speaking, these connections are giving between 1 and 4%
> packet loss.
> 
> Therefore, 3 questions: 1) is this level of packet loss likely to
> have the effect we're seeing?
I don't think it's the packet loss per se; it's more likely to be
jitter, and no, correcting for jitter in Asterisk isn't likely to make
much difference, because the jitter on upstream connections can be so
big that it overwhelms the jitter buffer.

This is a long way of saying that the effort you expend trying to fix
this problem isn't likely to help you much. Try a few things, but don't
spend too much money on it. What will solve the problem is better
Internet connections, and if you can't get those, you might well be stuck.

This is why I'm not in a hurry to recommend roaming to my customers
anytime soon.

-Stephen-

On Mon, 23 Apr 2007 14:05:55 +0100, Chris Bagnall wrote:
>Greetings list,
>Hoping someone might have experience with poorly-performing net connections
and which devices work best over them.
>One of our clients has a number of employees that work from home, and are
given a SIP phone to take with them and hook up to their broadband. For the most
part, this works fine, but
there are an increasing number where sound quality is poor ("chops" in
and out, generally only noticeable to the listener at the other end, not the
employee). Logic suggests it's an upstream
bandwidth issue, so we asked them to try when all other devices were turned off
(to cut out the "kids using bitTorrent" issues), but even with the
phone the only device, call quality was still
poor.
>Since the connections aren't paid for by the client, we aren't in a
position to mandate particular providers or speeds, but in each case, the
minimum was a 1mb/256k up ADSL. We asked
the employees to run some speed tests to determine real-world speeds, and in
each case upstream was around 220-235k (a little off the "official
speed" but not bad). Certainly way more
than the ~35kbps necessary for a g729 call, even with packet overheads.
>We've also tested the connections with a constant ping, and latency for
nearly all of them is sub-35ms.
>So, that leads me towards packet loss as the only thing left. Generally
speaking, these connections are giving between 1 and 4% packet loss.
>Therefore, 3 questions:
>1) is this level of packet loss likely to have the effect we're seeing?
>2) If so, are there any phones people have tried with particularly good
jitter buffering? If not, any ideas what else might be causing the issue.
>3) are some codecs naturally more "tolerant" of jitter than
others? i.e. would there be an advantage to using something apart from g729, and
if so, what would you recommend?

Chris,

The others responding on-list are certainly giving you good advice. I expect
that what you are suffering is unmanaged QoS at the roaming users end. This
almost certainly will be an issue
with 256k outbound on a network connection that is not dedicated to the voip
application alone.

Consider that companies like Packet8 or Vonage will sell their voip service to
these users, and generally make it work pretty well. They do it by providing the
a client side access device that
get inserted into the between the rest of the LAN and the DSL/cable modem. It
provides the bandwidth management to ensure workable voip.

Using a compressed codec like G729 or ILBC helps as well, but having a router
capable of QoS at each location is an absolute necessity. I prefer m0n0wall on a
Soekris Net4501. Others like
third party firmware on Linksys WRT devices....a little bit cheaper but less
professional IMHO.

Michael