[asterisk-users] SIP devices with packet loss tolerance

[Stephen Bosch]

Eric "ManxPower" Wieling 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 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.
> 
> PSTN <-> Asterisk <-> Internet <-> SIP Phone.
> 
> If the person on the PSTN side is having audio quality problems then the
> issue is not with the jitter buffer on the phone.  The problem in this
> case is the jitter buffer in Asterisk.  SIP is a signalling protocol.
> Audio is sent using the RTP protocol.  In versions of Asterisk before
> 1.4 there was no RTP jitter buffer in Asterisk.
> 
> Lack of an RTP jitter buffer in Asteirsk is why none of my clients have
> deployed phones off the corporate network.
> 
> If the person on the SIP phone side is having audio problems (not the
> case if I read your message correctly) then you have to look at the
> jitter buffer settings on the phone.
> 
> Remember jitter buffers (and QoS actually) is only applied to and is
> only effective for INCOMING traffic.
> 
> Yes, applying QoS to the outbound traffic of the internal interface of
> your router can give the illusion of limited QoS.  This happens because
> of the nature of TCP and will do nothing for non-TCP traffic.
> 
> Jitter is not the packet latency, but of the VARIANCE in latency.  Also,
> dejittering audio requires buffering and this buffering adds to the
> audio latency.  If you had a jitter buffer that could handle 3000ms of
> jitter (on a HughesNet satellite connection, for example) your audio
> would generally be great, the tradeoff is that you have just added 3
> seconds of latency to your audio and in anyone's book that sucks.

<Applause>

-Stephen-