Hi,
Sorry if this has been already asked/answered. I searched and found a few posts but not exactly like my setup.
I have my asterisk running on my router/gateway and the SIP client (Sipura) is behind the router (NAT). Is it possible to have RTP media traffic not go through Asterisk ?
I’m trying to get it to work with SIPPhone and FWD. I guess both support G711u, which is the default in my Sipura too.
Sipura on LAN <--------> NAT/router/asterisk <-------> SIPPhone or FWD
Here are the things I tried:
- canreinvite=yes
- no “t” or “T” in Dial() command
- forward Sipura RTP ports to Sipura on the router
Can someone help me with the settings on Asterisk and Sipura to get RTP traffic going directly and not through Asterisk ?
Thanks in advance.
it’s going to be a bit difficult behind NAT to get the invite to go to the Sipura if they are both sharing the same ip address I think. Maybe if you put the Sipura in the DMZ? I haven’t tried it. But - is there a big reason you need to do this? If you are coming off the internet, rtp passing though Asterisk isn’t really going to make a difference compared to the rest of the path it needs to go through.
p
p_lindheimer,
The reason I’m trying to offload RTP media traffic is because Asterisk is running on my router and the router doesn’t have CPU power to handle this too, on top of other things its doing. I know RTP traffic is anyways going to pass through the router, but when it has to go through Asterisk too, its becoming a bit too much for the little guy to handle. For anyone interested, I’m running Asterisk on OpenWRT running on my WRT54G V3.0.
I had it go through Asterisk and I was able to see the quality was not as good as Sipura directly registering with SIPPhone/FWD. I derived using Asterisk just for signaling and offloading RTP traffic would give back the quality.