Re: [AFMUG] DiffServ and the internet

[Adam Moffett via Af]

So one simple workaround I saw was was to match UDP packets at specific 
sizes.  Every single G.711 RTP packet my phone sends is a UDP packet 
that's 200 bytes every time.  I'm sure it would eventually overmatch 
something, but it's simple and low cost.   The example I saw was 
matching every UDP packet from 100-400 bytes that wasn't already matched 
by some other criteria.  A quick test here worked with a rule set to 
match at exactly 200 bytes....so I'm thinking rather than saying every 
mid size UDP packet is VoIP that maybe I'll match the specific sizes of 
packets in common codecs with 20ms frame sizes.

Yeah, it's more complicated than just giving the PBX a public and 
setting DSCP on every packet destined to that IP. I can't have 
everything end up in the HP queue, that will create more problems than 
the one I'm trying to solve. I need to figure out a way to identify 
the traffic and mangle DSCP on at the edge routers.

On 9/30/2014 11:04 AM, Adam Moffett via Af wrote:

....but like George, I would also be interested in some sort of rule 
that would match RTP voice traffic.   I don't see any easy way to do 
it, but wireshark seems to pick up on it reliably, so I guess there's 
a way.


+1
Seems like the easiest answer.

On 9/30/2014 11:05 AM, Ken Hohhof via Af wrote:
Is the company’s PBX behind their firewall?  If you give it a 
dedicated IP address, then you can tag based on destination IP.
*From:* George Skorup (Cyber Broadcasting) via Af 
<mailto:af@afmug.com>
*Sent:* Tuesday, September 30, 2014 9:55 AM
*To:* af@afmug.com <mailto:af@afmug.com>
*Subject:* Re: [AFMUG] DiffServ and the internet
These are business customers with on-site PBXs with a VoIP 
Innovations SIP trunk. Yeah, if we were running a local switch, 
then this problem would be a whole lot easier to solve, but that's 
not what I have to work with at this point.

As far as I can tell, there's no easy way to identify the VoIP 
Innovations audio streams. They come from tons of different source 
address and use random ports from 10000 to 20000. And it's a mix of 
G711 and G729.

No idea, I'm just the network guy.

On 9/30/2014 9:28 AM, Ken Hohhof via Af wrote:
That’s what I do, there’s another way?  We put customer ATAs on 
private IPs so it wouldn’t work if traffic bypassed our server.
Is there a configuration parameter on the SIP trunk that tells it 
to send RTP traffic directly to the endpoint?
We also have a multisite business customer that uses a hosted VoIP 
service (Star2Star) with an appliance at each site, we give each 
appliance its own public IP and tag traffic to those IPs.
*From:* Adam Moffett via Af <mailto:af@afmug.com>
*Sent:* Tuesday, September 30, 2014 9:03 AM
*To:* af@afmug.com <mailto:af@afmug.com>
*Subject:* Re: [AFMUG] DiffServ and the internet
I've been cheating up until this point.  If you force the audio to 
be bridged through your own server then you can tag all the 
traffic that goes to and from that server.  It doesn't seem to 
make a huge difference versus having RTP go straight to the 
carrier.  If you're not transcoding then the added CPU usage is 
minimal.  Faxing seems to work better if I'm not bridging the 
audio, but why am I faxing anyway, right?

I tried all kinds of stuff tonight, none were any good. I wonder 
if there's a way on MT to snoop SIP messages and look for the SIP 
contact IPs and mark those. Seems tricky. And I R no smrt enuf.

On 9/29/2014 9:37 PM, Chris Fabien via Af wrote:
Packet size and rate is pretty consistent right? Just a thought...
On Mon, Sep 29, 2014 at 8:05 PM, George Skorup (Cyber 
Broadcasting) via Af <af@afmug.com <mailto:af@afmug.com>> wrote:

    Speaking of DSCP and carriers zeroing it in the middle, I
    have some VoIP Innovations trunks. I know where the SIP
    messages are coming from, so I can mangle a DSCP value back
    onto those packets at ingress. But the RTP traffic comes
    from all over the freakin place, tons of different source
    address, never the same. I've asked if they could provide a
    list and pretty much got a no.

    Anybody have any ideas? Any way for a MT to identify an RTP
    stream and then dynamically add a mangle rule to change the
    DSCP value? My MT script-fu is not strong.