One way audio ASA

Unanswered Question
May 20th, 2009

Our setup is this. Call comes in via SIP Trunk, routes to CallManager through ASA and back out through to the remote site via VPN. Yeah, it's a complex setup I know.

The problem is that when a caller calls and gets connected with a user, the calling party can hear what the IP phone user is saying but the IP phone user cannot hear anything. If you look at the phone for sending/receiving packets, it is sending but receiving packets are not incrementing. This will happen intermittently (mostly about 75% of the time). The caller will get connected, everything will be working and then the other side goes silent.

My policy map is allowing sip across and the ACL is allowing everything across from the voice subnet. The phones never lose connectivity. I can ping fine from the voice subnet to the remote phone side.

policy-map global_policy

class inspection_default

inspect dns migrated_dns_map_1

inspect ftp

inspect h323 h225

inspect h323 ras

inspect rsh

inspect rtsp

inspect esmtp

inspect sqlnet

inspect skinny

inspect sunrpc

inspect xdmcp

inspect sip

inspect netbios

inspect tftp

inspect icmp

inspect icmp error

Both ASA's are running 8.0(4)29. Any suggestions?

I have this problem too.
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BrinksArgentina Fri, 05/22/2009 - 09:25

Were call come from?

Another IP phone registered on another callmanager?

You must go to de basis of the problem capturing ip traffic in both endpoints. Check source/destination address and check all the path.

Post all information that you can and I will try to help you. I have the same problem a year ago and was a BGP policy routing problem.


Guido.

Please rate all the helpful comments.

I would make sure that there isn't any other NAT or ACL's in line of the receive traffic. Then I would take two old pc's and install Wireshark. Using TSHark, capture the SIP, RTP and ICMP traffic into a handful of ring buffer files. When the problem occurs, see if the audio streams are present on the the other side of the ASA. If not, you will want to keep moving toward the other end.

Additionally, you might want to see if your ASA is logging any dropped packets from the other end. In any case Wireshark is a great tool for working with SIP. You can actually listen in the RTP Streams they are using G711.

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