We're having voice problem issues with calls to the PSTN when two people are talking at the same time. We've been through a couple of weeks troubleshooting with TAC to no resolve. Has anyone else experienced similar issues?
One-way audio occurs when one person cannot hear another person during a call. This can be caused by an improperly-configured Cisco IOS gateway, a firewall, or a routing or default gateway problem, among other things.
There are a number of causes for one-way audio or no audio during a call. The most common cause is an improperly-configured device. For instance, Cisco CallManager handles the call setup for a Cisco IP phone. The actual audio stream occurs between the two Cisco IP phones (or between the Cisco IP phone and a gateway). So, it is entirely possible that the Cisco CallManager is able to signal to a destination phone (making it ring) when the phone originating the call does not have an IP route to the destination phone. This commonly occurs when the default gateway in the phone is improperly configured (manually or on the Dynamic Host Configuration Protocol (DHCP) server).
If a call consistently has one-way audio, try to ping the destination Cisco IP phone using a PC that is on the same subnet as the phone and has the same default gateway. Take a PC that is on the same subnet as the destination phone (with the same default gateway as the destination phone) and ping the source phone. Both of those tests should work. Audio traffic can also be affected by a firewall or packet filter (such as access lists on a router) that may be blocking the audio in one or both directions. If the one-way audio occurs only through a voice-enabled Cisco IOS gateway, check the configuration carefully. IP routing must be enabled (examine the configuration to make sure that "no ip routing" is not found near the beginning of the configuration). Also, if you're using RTP header compression to save bandwidth across the WAN, make sure that it is enabled on each router carrying voice traffic that attaches to the WAN circuit. There should not be a situation where the RTP header is compressed on one end but cannot be de-compressed on the other side of the WAN. A sniffer is a very useful tool when troubleshooting one-way audio problems because you can verify that the phone or gateway is actually sending or receiving packets. Diagnostic CDRs are useful in determining if a call is experiencing one-way audio because they log transmitted and received packets (refer to Lost or Distorted Audio). You can also press the i button twice (quickly) on a Cisco IP Phone 7960 during an active call to view details about transmitted and received packets.
Note: When a call is muted (mute button pressed on a phone), no packets will be transmitted. The Hold button stops the audio stream, so no packets are sent in either direction. When the Hold button is released, all the packet counters are reset. Remember that Silence Suppression must be disabled on both devices for the TX and RX counters to stay equal. Disabling Silence Suppression system-wide will not affect Cisco IOS Gateways.
MTP and One-Way Audio
If you are using Media Termination Point (MTP) in a call (to support supplementary services such as hold and transfer with H.323 devices that do not support H.323 version 2), check to see if the MTP allocated is working correctly. Cisco IOS routers support H.323 version 2 beginning in release 11.3(9)NA and 12.0(3)T. Starting with Cisco IOS release 12.0(7)T, the optional H.323 Open/Close LogicalChannel is supported, so that software-based MTP is no longer required for supplementary services.
The MTP device, as well as Conference Bridge and Transcoder, will bridge two or more audio streams. If the MTP, Conference Bridge, or Transcoder is not working properly, one-way audio or audio loss might be experienced. Shut down MTP to find out if MTP is causing the problem.
Not sure if this is the same issue, but we have discovered with the TAC that the speakerphone on the 7960/7940 does not duplex. If you have the phone on speakerphone, the other party speaks, you cannot talk. This happen in the network and out of the network (local and through PSTN). 7935's have no issue. It may be an echo cancel issue on the phone setup. We are on CM 3.1.2 C and have tried SPA, B and phone loads. Handset is fine. The TAC can produce the problem every time in their lab, word is that it may be a bug and was sent over to BU for development. At this point we have considered going to RadioShack for standard speakerphones and sending through FXs. So much for polycom technology!
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