ONE WAY AUDIO SIP TRUNK

[kennedymacharia]

Hi all I have a sip trunk between CUCM and an CME. That was working okay untill  the Firewall at the branch router where the cucm is located was replaced with a cisco router. now there is only one way audio. I have bind commands on the dial peer at the cucm as follows

dial-peer voice 104 voip
corlist outgoing VOIP
description **Incoming Call from CUCM Trunk**
session protocol sipv2
session target ipv4:172.27.199.11
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1.110
voice-class sip bind media source-interface GigabitEthernet0/1.110
dtmf-relay h245-alphanumeric

What could be my problem.?

[Brandon Buffin]

Sounds like the new firewall is blocking traffic. Make sure it is configured to allow ports 5060/5061 between CUCM and CME in both directions. Also, could the firewall be doing NAT? NAT can break SIP traffic.

Brandon

[Jonathan Schulenberg]

If the call is connecting but there is one-way audio then CUCM and the CME GigabitEthernet0/1.110 interface can route to one another. You would need to look at the IP addresses from the SDP media negotiation and whether they match what you expect. I agree with Brandon that a firewall (default port range of UDP 16384-32768) or NAT could be blocking the RTP traffic but it's also possible that one endpoint cannot route to the other's IP address.