10-05-2009 07:27 AM - edited 03-15-2019 07:54 PM
Hi
We are implementing QoS for IP Telephony infrastructure. The IP Phones mark the packets for signalling and RTP. This would be for the traffic originating at the phone towards the voice gateway.
What and how should the marking happen for a call from PSTN towards the Ip phone? Is there separate marking to be done on the voice gateway or the switches.
Thanks.
10-05-2009 02:25 PM
Router sets the right value by default. With H.323 or SIP, you cahnge it under dial/peer. With MGCP, configure the CM for this.
10-05-2009 05:38 PM
what ever is doing the marking for RTP traffic coming from the Voice gateway:
add the following line to the access list that defines the required DSCP or presedence value:
permit udp any any range 16384 32767
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