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.
