[Asterisk-Users] QoS for improvements

[Michael Graves]

Jean-Christophe,

Thank you for the explanation. I've never been in a situation demanding
adjustment of MTU. It's not so much that I think I have a better way,
only that my circumstances lend themselves to a simple solution. I did
start out using * with only 256k upload speed. I decided to stay with
G.711 and purchase the better connection, since it was available. 

In your area where raw bandwidth is costly is there any sense in using
ISDN lines instead of ADSL? I'd love to dump my SBC POTS lines and get
two BRIs, but BRI capable hardware meeting US standards is
scarce/non-existent.

Michael

On Fri, 06 May 2005 18:34:00 +0200, Jean-Christophe Heger wrote:

>That's funny, people having good bandwidth always have a better way to
>do it. You should feel lucky, because no one provides 768kbps upstreams
>in Switzerland, except if you want to pay 1'000 USD per month for a
>leased line.
>
>There is nothing complicated, just mathematics.
>
>Here is the formula:
>
>MTU: Maximum transmit unit = 1492 Bytes (ADSL)
>UP: Up stream
>t: time spent for a full framed packet (1492 Bytes)
>
>t = 8 * UP / MTU
>
>128k upstream -> 91 ms
>256k upstream -> 45 ms
>512k upstream -> 23 ms
>768k upstream -> 15 ms
>
>Using a codec, such as GSM or G.729, will take around 20 to 30 ms for
>encoding and decoding.
>
>While you wait for a full framed packet to go through the ADSL line, a
>voip packet will wait need 20 ms for encoding, 91 ms for waiting (at the
>most), 30 ms to go to the destination (at the best), and 10 ms to be
>decoded. Total: 20+91+30+20 = 161 ms. Even worse, the jitter balance
>will play around 100 ms, because of waiting on full framed packed.
>That's what I call "breaking the jitter", because not all equipment does
>support such jitters.
>
>Depending on the line and the distance (hops), you can easily add 50 ms,
>bringing the total around 200 ms. Therefore we consider a conversation
>as good, when the overall delay is under 150 ms. Bringing the MTU to 700
>ms does bring back the overall delay to this target, and the jitter to
>50 ms.
>
>Regarding the results, 768 kbps up stream is working even without QoS (<
>100 ms). PFIFO (TOS=lowdelay) is good enough for perfect communications.
>
>So, any other magical solution ?
>
>Jean-Christophe
>
>
>Michael Graves a écrit :
>
>>Sometimes this all sounds so complicated....but it needn't be. I
>>suppose it can vary with the size of your installation.
>>
>>I use a m0n0wall router on a Covad DSL line. Using m0n0's traffic
>>shaping feature I establish inbound and outbound pipes which are
>>bandwidth restricted to just less than my mesured average DSL rate.  I
>>then break my traffic into three priority ques in each direction;
>>highest priority, medium priority, low priority.
>>
>>I assign all IAX traffic in/out to the highest priority que, and map
>>all IAX ports to the * server inside the LAN.
>>
>>In fact, I just used the Magic Shaper Wizard in m0n0, then added IAX
>>specific entries to give it highest priority. The whole process took
>>about a half hour. Just as easy as the Linksys BEFSR-81 that I had
>>before, but more reliable and more controllable.
>>
>>Now to be fair, I only have about 6 phones. I use IAX2 to all my ITSPs
>>and SIP in-house only. My DSL is 3M down / 768k up.
>>
>>Michael
>>
>>On Fri, 06 May 2005 08:37:20 +0200, Jean-Christophe Heger wrote:
>>  
>>
>
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--
Michael Graves                           mgraves at pixelpower.com
Sr. Product Specialist                          www.pixelpower.com
Pixel Power Inc.                                 mgraves at mstvp.com

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