[asterisk-users] SIP devices with packet loss tolerance

Nicholas Campion campnic at gmail.com
Mon Apr 23 07:25:46 MST 2007

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 at 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
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