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

[Andrew Gillham]

On Mon, May 12, 2003 at 08:35:46PM -0500, Eric Wieling wrote:
> 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.

You can lower your maximum receive window size and your maximum segment
size so the email machine is not able to use 144Kbit/s.

If you can use your Linux box as a port forwarder you can set an explicit
route for your mail server with 'window <value>' and 'mss <value>' so email
has a window size of 1460 or something.  Or a higher window size and a lower
mss.  The Linux box has to initiate the connection, hence the need for
a port forwarder / proxy, not just NAT or routing through it.

This is just a hack, but would force TCP connections to your ISP to
run slower.

Otherwise FreeBSD has the 'dummynet' feature that will do a certain amount
of traffic shaping.  NetBSD (and OpenBSD) have ALTQ for similar features.

Obviously you really want QoS so you can run at full speed and have your
ISP router handle congestion.

Meanwhile, I was talking to a buddy SIP phone to SIP phone over an IPSEC
tunnel.  RTT for ping was about 600ms due to congestion on his cable modem
provider's network, but our voice call was perfectly fine.  It definitely
did not have 600ms of latency!  The Cisco 7960 will set the TOS bits based
on a configuration option.

So some ISPs apparently support QoS without really advertising it.

-Andrew