[asterisk-users] SIP devices with packet loss tolerance

[Stephen Bosch]

Hi again:

Michael Graves wrote:
> 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.

If this were indeed the cause of the problem, then it would have
resolved by simply connecting the SIP phone directly to the DSL modem.
In that case, the *only* traffic going out is voice traffic. That's
really all that Vonage ATA is doing -- making sure that the voice
traffic gets preferential treatment on its way out. I know enough Vonage
users who get crap call quality anyway, outbound QoS or not. What Vonage
is doing is playing the odds; they're betting that enough people will
have adequate broadband connections to make the enterprise worthwhile.

Anyway, if that's all that were needed, the cheaper way to accomplish it
would be to plug the rest of the roaming user's network into the LAN
port on the back of the SIP phone. You get some limited traffic
prioritization there for the cost of admission. Chris has already tried
that.

QoS is meaningless unless the ISP is supporting it (and, ideally, every
network device along the patch between Chris' Asterisk system and the
roaming users). In general, QoS as a notion sounds exciting and very
cool, but who can implement it? The only ones really benefiting from it
so far are large corporate users with their own WANs who are
implementing internal VoIP over their entire business. I can think of a
few American investment banks. Yes, there are some ISPs that are
offering QoS to their customers (Shaw in Canada comes to mind), but if
you think that comes for free, well... Shaw charges $15/month for
residential QoS (that is, unless you are buying *their* VoIP service).
At that price, I'll keep my PSTN phone, thanks.

I would bet money that these users would have just as much trouble with
a Packet8 or Vonage device.

Someday, we might see QoS of some kind over all the public Internet.
Someday long into the future. I don't think it will come for free.

> 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.

Again -- in the circumstances described above, it is utterly meaningless
unless the devices in the path support it also. As for the codecs --
compressed codecs are great for reducing the average bandwidth
requirement but do nothing for latency.

I say again -- it is wasted effort. Try pounding the pavement for better
broadband connections for these users.

-Stephen-