[asterisk-users] SIP devices with packet loss tolerance
dickson at covad.net
Mon Apr 23 18:12:43 MST 2007
On Mon, 23 Apr 2007 14:05:55 +0100, Chris Bagnall wrote:
>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
>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?
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.
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