[pfSense Support] naive prioritization of VoIP?

2011-06-02 Thread Adam Thompson
I’m trying to make sure VoIP has the best possible quality for a small amount 
of effort.

I still don’t understand QoS, even the wizard is baffling to me – for whatever 
reason QoS is a layer my brain just doesn’t want to accept.

What I’ve done in the past on other firewalls is a trivial “priority” setting: 
without configuring any queues, buckets, shapers, etc., I would simply create a 
rule matching SIP traffic (either by port, by NBAR-ish/L7 application or by IP 
address) and set the “priority” to “high”.  I really have no idea what that 
does under the hood, whether on FortiNet, Cisco,  or PaloAlto.

Is there anything that simple that I can do under pfSense?

 

Thanks,

 

-Adam Thompson

athom...@athompso.net

 



Re: [pfSense Support] naive prioritization of VoIP?

2011-06-02 Thread karlfife

I've had great luck with VoIP and pfSense.
To be clear, there's no such thing as 'real' end-to-end guarantee of 
quality of service unless you're talking about MPLS or similar 
technologies.  What you want is called 'traffic shaping'


For ordinary people with ordinary connections, the idea is as follows:
PART 1 -
Starve the pipe!
You must utilize your internet connection below its maximum 'guaranteed' 
throughput, otherwise you will have no control over the upstream buffers 
(see buffer bloat), and your real-time application, VoIP or otherwise 
will suffer.  In VoIP, that means that packets will either not arrive, 
or arrive so late as to exceed the VoIP UA's jitter buffer, and will 
result in subjective quality factors, technically referred to as Shitty 
quality (Drops, stutters, etc).

PART 2 -
Prioritize your real-time packets!
Now that your pipe is VERY SLIGHTLY underutilized, you have left 
yourself the ability to instantly insert the VERY NEXT VoIP packet into 
your data stream if one should happen to arrive (the very NEXT VoIP 
packet is the one you have to be preemptively ready for).  When that 
packet arrives, the 'shaper' immediately adjusts TX/RX rates to CONTINUE 
to keep the pipe slightly underutilized.  This is why you need to know 
your up/downstream speeds to configure your traffic shaper.  All of the 
NON real-time stuff can be put 'in line'.  All of that lower-priority 
stuff essentially must 'wait in line' to get IN or OUT, at that magic 
rate JUST UNDER the maximum rate to keep the pipe CONSTANTLY SLIGHTLY 
UNDERUTILIZED. Naturally VoIP packets gets to go to the front of the 
line in inbound or outbound queue.


That's pretty much it. The 'starve the pipe' business is why it's not as 
simple as Simply prioritizing Voip


PFSense makes it quite simple however.  Just measure your link speed at 
something like speedtest.speakeasy.net.  Walk through the traffic 
shaper wizard specifiying that VoIP gets top priority, whether that's 
the internal IP address (or alias) of your VoIP ATA, Astrisk server or 
VoIP telephone.


Good luck
-Karl







On 6/2/2011 4:03 PM, Adam Thompson wrote:

I’m trying to make sure VoIP has the best possible quality for a small
amount of effort.

I still don’t understand QoS, even the wizard is baffling to me – for
whatever reason QoS is a layer my brain just doesn’t want to accept.

What I’ve done in the past on other firewalls is a trivial “priority”
setting: without configuring any queues, buckets, shapers, etc., I would
simply create a rule matching SIP traffic (either by port, by
NBAR-ish/L7 application or by IP address) and set the “priority” to
“high”.  I really have no idea what that does under the hood, whether on
FortiNet, Cisco,  or PaloAlto.

Is there anything that simple that I can do under pfSense?

Thanks,

-Adam Thompson

athom...@athompso.net mailto:athom...@athompso.net



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Re: [pfSense Support] naive prioritization of VoIP?

2011-06-02 Thread Adam Thompson
This begs the question of what, exactly  do all those other firewalls DO when I 
set priority.
...speaking of VoIP, does anyone know if the FreeSwitch packages are ever 
getting updated?  Or if the -dev version really is following HEAD?
-Adam


karlf...@gmail.com wrote:

I've had great luck with VoIP and pfSense.
To be clear, there's no such thing as 'real' end-to-end guarantee of 
quality of service unless you're talking about MPLS or similar 
technologies.  What you want is called 'traffic shaping'

For ordinary people with ordinary connections, the idea is as follows:
PART 1 -
Starve the pipe!
You must utilize your internet connection below its maximum 'guaranteed' 
throughput, otherwise you will have no control over the upstream buffers 
(see buffer bloat), and your real-time application, VoIP or otherwise 
will suffer.  In VoIP, that means that packets will either not arrive, 
or arrive so late as to exceed the VoIP UA's jitter buffer, and will 
result in subjective quality factors, technically referred to as Shitty 
quality (Drops, stutters, etc).
PART 2 -
Prioritize your real-time packets!
Now that your pipe is VERY SLIGHTLY underutilized, you have left 
yourself the ability to instantly insert the VERY NEXT VoIP packet into 
your data stream if one should happen to arrive (the very NEXT VoIP 
packet is the one you have to be preemptively ready for).  When that 
packet arrives, the 'shaper' immediately adjusts TX/RX rates to CONTINUE 
to keep the pipe slightly underutilized.  This is why you need to know 
your up/downstream speeds to configure your traffic shaper.  All of the 
NON real-time stuff can be put 'in line'.  All of that lower-priority 
stuff essentially must 'wait in line' to get IN or OUT, at that magic 
rate JUST UNDER the maximum rate to keep the pipe CONSTANTLY SLIGHTLY 
UNDERUTILIZED. Naturally VoIP packets gets to go to the front of the 
line in inbound or outbound queue.

That's pretty much it. The 'starve the pipe' business is why it's not as 
simple as Simply prioritizing Voip

PFSense makes it quite simple however.  Just measure your link speed at 
something like speedtest.speakeasy.net.  Walk through the traffic 
shaper wizard specifiying that VoIP gets top priority, whether that's 
the internal IP address (or alias) of your VoIP ATA, Astrisk server or 
VoIP telephone.

Good luck
-Karl







On 6/2/2011 4:03 PM, Adam Thompson wrote:
 I’m trying to make sure VoIP has the best possible quality for a small
 amount of effort.

 I still don’t understand QoS, even the wizard is baffling to me – for
 whatever reason QoS is a layer my brain just doesn’t want to accept.

 What I’ve done in the past on other firewalls is a trivial “priority”
 setting: without configuring any queues, buckets, shapers, etc., I would
 simply create a rule matching SIP traffic (either by port, by
 NBAR-ish/L7 application or by IP address) and set the “priority” to
 “high”.  I really have no idea what that does under the hood, whether on
 FortiNet, Cisco,  or PaloAlto.

 Is there anything that simple that I can do under pfSense?

 Thanks,

 -Adam Thompson

 athom...@athompso.net mailto:athom...@athompso.net


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Re: [pfSense Support] naive prioritization of VoIP?

2011-06-02 Thread Chris Buechler
On Thu, Jun 2, 2011 at 6:12 PM, Adam Thompson athom...@athompso.net wrote:
 This begs the question of what, exactly  do all those other firewalls DO when 
 I set priority.

It varies, but generally there's more to it than setting priority. You
need link speed as well as you need the firewall do to the queuing and
never hit any queuing on your modem or elsewhere outside your control,
otherwise priority is useless as you're just going to blast traffic
out to be queued where your ISP is rate limiting you.

 ...speaking of VoIP, does anyone know if the FreeSwitch packages are ever 
 getting updated?

It won't be, it's going to be replaced with FusionPBX, the successor
to our Freeswitch package. Mark Crane created it, then moved on to
FusionPBX, and will be creating a package for it to replace the
Freeswitch package at some point (not sure of his exact plans).

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