VoIP (over frame) - no sound in either direction after call is established.

[adam.wilkins]

I have two 3640 routers connected via a frame-relay cloud (a router acting as a frame-relay switch). Each 3640 has a 2FXS WIC card. I can make calls all day - the other phone rings, I pick up, but I cannot hear myself speak through either phone. Am I doing something wrong here???? Below are the basic configs. And, I can ping back and forth all day and ip routing is enabled - I have BGP running and peers established (I read somewhere else that "ip routing" was an issue in cases like this - I just wanted to state that to rule that out)....

Router A

interface Serial0/1

encapsulation frame-relay

!

interface Serial0/1.1

ip address 150.150.10.1 255.255.255.252

frame-relay interface-dlci 100

frame-relay ip rtp header-compression

!

dial-peer voice 11 pots

destination-pattern 1111

port 3/0/0

!

dial-peer voice 2222 voip

destination-pattern 2222

session-target ipv4:150.150.10.2

ip precedence 5

Router B

interface Serial0/1

encapsulation frame-relay

!

interface Serial0/1.1

ip address 150.150.10.2 255.255.255.252

frame-relay interface-dlci 100

frame-relay ip rtp header-compression

!

dial-peer voice 22 pots

destination-pattern 2222

port 3/0/0

!

dial-peer voice 1111 voip

destination-pattern 2222

session-target ipv4:150.150.10.1

ip precedence 5

[Niels Brunsgaard]

When a call is active try using thjs command on both sides to verify what IP addresses are being used for the RTP stream:

BrisVG200gwy01#sh call active voice | include Remote

RemoteIPAddress=10.64.40.194

RemoteUDPPort=17598

RemoteSignallingIPAddress=10.64.40.194

RemoteSignallingPort=1391

RemoteMediaIPAddress=10.64.102.86

RemoteMediaPort=17598

BrisVG200gwy01#

You may need to use the interface command 'h323-gateway voip bind srcaddr' to bind the H323 stack to a certain IP address, and then point the dialpeers to that IP address.

[fbahram]

First if you have a very low bandwidth and obviously no traffic shaping, you will get very poor quality voice if data is running on the link. But if we are getting absolutely no audio, then try the following:

1. On both routers, make sure when a call is up, the 'show call active voice' output shows active/active

2. Show frame ip rtp header (done a few times) should show both transmit & received compressed packets incrementing. you can clear it first and then look at it w/ a new call (clear ip rtp header)

3. Make sure you have 'voice rtp sent-recv' and 'voice call send-alert' on both routers, globally defined

4. Make sure 'show voice dsp' shows the tx/rx packets in and out of dsps incrementing for the BUSY DSP channel on which you have the call.

5. show ip socket=> the ip addr for local and remote for the voip ports should match your session target ip addresses.

Hope this helps.

[adam.wilkins]

Thanks for the help...

I should need no compression or anything. This is in a lab environment and I have a full T1 frame-relay between the two routers. No other traffic is traversing this link.

I will try it out though.

Thanks..