Thanks for your reply. Yes, I have access-lists. I didn't even think of that for some reason. Which ports are necessary for sound to travel properly?

Hi,

Cisco's standard VOIP ports are UDP 16384-32767.

You may find this document I wrote helpful for looking at the one-way audio issue with 'show call active voice brief'.

hth,

nick

I am suspecting a routing issue. I need a little more information. When I establish a call from my CIPC is the RTP stream going to the gateway directly or is it going to the CME which is redirecting it to the gateway?

If your CME is using a VOIP protocol to reach another gateway (H323/SIP), then your media will be terminated on CME.

You can check this on your computer fairly easily by installing wireshark.

For hard phones you can go to the website of the phone and it will tell you, but CIPC is on your PC so it's different.

-nick

Also CIPC has web server that gives you information.

Do you happen to know which port it listens on? I only show two ports open with TCPView. 56583 which is connected to CME at port 2000 and 58932 UDP which definitely isn't a webserver.

RTP ports are negotiated dynamically.

Browse to phone address or press ? twice rapidly to get informations about RTP.

I am talking about the web server that CIPC has. How do I access that?

I have verified connection between CME and the gateway connected to the PSTN. I setup an access-list on the CME interface:

permit ip host ip.add.of.gw host ip.add.of.cme

and I put an access-list on the gateway:

permit ip host ip.add.of.cme host ip.add.of.gw

The counters go up on both of them. Also, I can ping between the two without issue. Is there something beyond firewall issues that would cause no sound?

Alright... it was definitely a combination of firewalls and ACLs. Thanks for your help fellas.

You are welcome. Thanks for the nice rating and good luck!