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Cisco video RTP ports

We have a Call Manager deployment using CUCM 11.0.1. Our customer is using DX-650 endpoints in many VPN locations. And also are SX 80 codecs for video and they are also using CMR cloud. The dx650 and Jabber endpoints will be heavily used as video clients hence QOS is a must in the location. As our customer does not agree to use a separate voice subnet for IP phones we are using RTP ports to enforce markings on packets. In order to do so we need to know what are the separate ranges used by Voice media and Video media.

Can some one please help us with this.

1 Accepted Solution

Accepted Solutions

Jonathan Unger
Level 7
Level 7

Hi There,

The same protocol RTP (Real-time Transport Protocol) is used to carry Video and Voice, the port range for RTP is UDP 16384-32767.

Since the port range is pretty large, it isn't recommended to trust markings just based on the port number. If segmenting the network is not a possibility you might want to look into classifying packets using NBAR - explanation here

Another advantage of using NBAR would be you can differentiate between voice and video RTP streams (you generally do not want your video streams ending up in your EF queue).

Does that help?

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2 Replies 2

Jonathan Unger
Level 7
Level 7

Hi There,

The same protocol RTP (Real-time Transport Protocol) is used to carry Video and Voice, the port range for RTP is UDP 16384-32767.

Since the port range is pretty large, it isn't recommended to trust markings just based on the port number. If segmenting the network is not a possibility you might want to look into classifying packets using NBAR - explanation here

Another advantage of using NBAR would be you can differentiate between voice and video RTP streams (you generally do not want your video streams ending up in your EF queue).

Does that help?

As of CUCM 11.x we can customize the SIP profiles of SIP phones in order to use a separate RTP port range for Video. And we can also customize what the DSCP value should be for voice and video separately. We used this functionality in the sip profile to enable the QOS for our customer.