How to Isolate & Troubleshoot One Way Audio on VoIP

 In Troubleshooting and Performance Monitoring
Reading Time: 4 minutes

Anybody working with VoIP has come across the dreaded one-way audio complaint. As a matter of fact, it is the third most frequent problem related to VoIP/UC in modern day telephony network services. However, as soon as you have isolated exactly the root cause of the problem in your network, you can ensure that it never happens again.

One Way Audio Causes

The cause of one-way audio is often a defect in the Network Address Translation (NAT) function. NATting is usually performed as your IP traffic traverses the wide-area network and enters your private corporate network. In this article, we will explain NAT, where it occurs in your network, and why it breaks VoIP audio. We will show you how to analyze the problem in a SIP monitor and how to fix it.

Why do we need NAT?

As you know, in order to make the best use of the limited IP address space available worldwide, we translate the public IP address used to route to our Corporation network into a private IP address for used inside the internal corporate network to route to each workstation or desktop device by IP address. A few IPv4 IP addresses are identified for private networks. These are listed below. The remainder is public IP addresses and allocated to unique resources worldwide so that we can route to them over the Internet.So when an IP packet enters your corporate private network, the IP address is transposed by the NAT device to an internal one, specific to the Computer involved in the session

During the NAT process, the firewall/home-router CPE device changes only the destination address. the incoming or outgoing source IP address are never changed or NATted.

 Where in the network is the NAT function?

The NATting function may be performed on the home router of a remote worker. Or it might be performed in your firewall as traffic enters your corporate network. When performing Network Address Translation on VoIP packets, specific configurations must be implemented on the customer premises equipment (CPE) or in your network.

Your corporate firewall may include Application Layer Gateway (ALG) functionality and this needs to be correctly configured relating to VoIP/SIP. Some ALG functionality puts a huge strain on the resources of the firewall. Sometimes it’s better to turn off the ALG function and have the VoIP/SIP handled by the network element specifically designed to secure your voice and that is a Session Border Controller (SBC).

How does the SIP Protocol Manage Routing of the Audio Packets?

Let’s look at a typical inbound VoIP/SIP call to see what goes wrong to cause one-way audio.

Above you can see the SIP INVITE setting up the call as seen in the message flow diagram of the Oracle EOM service assurance platform.

Message 1 and message 2 are effectively the same INVITE of the same call, one from the public side of the NAT device and the second one from the private side of the NAT device.

The To field in the second INVITE is the Address of the Endpoint / Softphone after it has been NATted.

So, What is Actually Causing One Way Audio?

Messages 1 & 2 show the SIP INVITE packet incoming from the PSTN through the CPE NAT device. The SDP part of this INVITE instructs the receiving SIP endpoint/softphone to send it’s audio to the IP address given in the “C=” header of the SDP.

The response from the terminating software on MAC Wi-Fi in message 11 tells the other end where to route the RTP audio (route it to 192.168.100.100:UDP port 49922). The NAT protocol function operates only at the IP layer and changes the IP to the private address on the SIP Signaling packets. But to route the subsequent RTP multimedia, the NAT device needs to look inside the SIP packet which controls all aspects of the phone call including routing and features of the RTP media. Routers and firewalls do not typically do this. SIP is an application function so an ALG may be able to perform this function but often problems occur.

Unfortunately, the endpoint/softphone answering this call is at 192.168.1.123, not 192.168.1.100.  Hence the audio is being sent to the wrong IP address and is not received for the incoming direction, resulting in one-way audio.

Sometimes SIP packets can go through two customer premises devices affecting Double NATting. This frequently configures an incorrect destination for the media. The best way to remediate this is to eliminate the NAT process in one of the CPE devices.

4G/LTE networks use SIP in the core. NAT is also used at network interconnects so this troubleshooting method can be used for wireless troubleshooting as well as all SIP-based wired or wireline the IP networks end-to-end.

Preventing One-Way Audio From the Source

Problems with one-way audio can be isolated right down to the specific device and configuration of the device in a few minutes. If your network suffers from this problem from time to time and you want to be able to fix the problem forever, you need to start by isolating instances..

How often does one-way audio occur in the context of all other SIP related problems or trouble tickets?

If you’d like to equip your team with the skills and tools to fix all SIP problems within a few minutes, or a comprehensive service to isolate problem-causing devices in your network, get in touch with us as below.

For more information, view:
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