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 Sent: Tuesday, September 30, 2014 9:03 AM To: [email protected] 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 <[email protected]> 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.
