[Asterisk-Dev] Anyone doing QOS routing on Linux for SIP/RTP?

[John Todd]

As mentioned, if packet loss starts to happen then TCP will back off. 
That's the way it's designed.  UDP, however, will continue to stream 
at whatever rate your codecs are set.  In the long term, of course, 
this makes UDP very un-friendly traffic to co-exist with, and 
multiple UDP streams will just continue to hose the network down 
until unusable.  However, TCP has the pleasant feature of back-off, 
so TCP will just get slower, and slower, and slower but will (within 
reason) maintain state and continue to work.

I would strongly suggest picking up a copy of "Internetworking with 
TCP/IP" by Douglas Comer.  It is an excellent book for the beginning 
network admin, and an even better book for the very advanced network 
admin.


JT


>What I *really* want is QoS on my access line to my ISP.  I have a 144K
>SDSL (only thing available at my location) and I receive some fairly
>large e-mails at times.  I don't want congestion on my link between me
>and my ISP to cause problems with my VoIP calls.
>
>On Mon, 2003-05-12 at 20:12, Adam Goryachev wrote:
>>  > Are there ANY ISPs that support QoS on their backbones?
>>
>>  Well I re-sell NEC Nextep DSL services, while they don't support QoS across
>>  their links, you can buy a particular service level up to and including a
>>  'leased line equivalent' service. Of course, it is really a ATM network with
>>  DSL tails, so even the lowest service level, it is always first packet in =
>>  first packet out, so if you order the packets correctly at your end, they
>>  will arrive in the same order at the remote end. The lower service levels
>>  would mostly affect total available bandwidth mostly seen as jitter...
>>
>>  (I'm assuming, untested and I don't wholly understand very much about VoIP
>>  etc)
>>
>>  Regards,
>  > Adam
>>
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