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] <mailto:[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.
