I have a server running Asterisk and a Webserver.
I have force my SIP file (SIP.CONF) with TOS=0xb8 and and as far as I know, this is the same for RTP protocols…
Now, On this machine, I have Asterisk and a Webserver running.
on my router, I have prioritized, all UDP protocols ( DSCP=46 ) (which should correspond to a TOS=b8)… on the port [8500 - 31000], which are the ports I wish to use for RTP packets
[ I choose 8500, for the use of X-Lite software ]
But if somebody downloads something on the webserver… the communication is breaking … (upstream)
Any idea how to select RTP packet ONLY ?
Is there any other possibility to get a better “filter” ?
The voice quality may not be suffering because of the network congestion, it could be due to server load. I would HIGHLY suggest you move the webserver to another machine. Asterisk really should be on its own server with no other tasks to perform other than telephony.
To be honest, I just want to match the router config with the Asterisk config
I did change the web server -> a different machine. The problem is the same. So that seems NOT to come from the server.
What I did, is to force the SIP and RTP protocol to a specific TOS (0xb8)
On the Billion 7402G Router, I selected : IPv4 TOS Priority Control -> TOS 46 (which is I think the DSCP and not the TOS)
Prioritized the port range of Asterisk and select the IP address of the server…
I did run the WireShark, and I can see that the packet from the server are correctly marked : SIP -> DSCP : 0x2e ( =46)
But also : RTP packet : DSCP:0x2e (Packet OUT)
So that seems to be OK… but that’s not… And I have no idea why
