Yeah....if the ATA had a public IP, or was on NAT behind a router with a
public IP, then the RTP traffic could go directly between the ATA and
the carrier.
If you're using asterisk, the old (1.2 and 1.4) paramter is
canreinvite=. In 1.6+ they renamed it to directmedia=
If canreinvite=yes or directmedia=yes then the endpoints are allowed to
bypass asterisk and talk to each other directly. If they don't have
direct connectivity to each other, then obviously they can't and won't
do that no matter what you set the option to.
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:[email protected]>
*Sent:* Tuesday, September 30, 2014 9:03 AM
*To:* [email protected] <mailto:[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] <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.
