Qos & tos


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” ?

I am using a Billion 7402G Modem Router


Any news on this stephane?

Do you want to change the prioritisation in the asterisk configuration files or is this your router’s configuration that you need(ed) to change?


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 :frowning:

On thing to check is you asterisk log files and check that tos-0xb8 is being set. you may find in the log an error saying it cant set it.

worth checking. Ian