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Last updated on 2025年2月28日

How can you troubleshoot one-way audio in a VoIP call?

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One-way audio is a common problem in Voice over IP (VoIP) calls, where one party can hear the other, but not vice versa. This can be frustrating and affect the quality of communication. Fortunately, there are some steps you can take to troubleshoot and resolve this issue. In this article, you will learn how to identify the possible causes of one-way audio, how to use diagnostic tools and commands, and how to apply some solutions based on different scenarios.

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Christina Edenfield

Passionate DevOps Engineer | AWS Certified, Automation
Leonardo Quintero

PEng. | DevOps | Network Automation | VoIP | Always looking to work smarter, not harder
Abin Mathew, P.Eng

Carrier Voice Networks, VoIP, Contact Center | Ribbon, Cisco, Metaswitch, Nokia | SIP, RTP, SS7, H.248 | Solution…

1 Check the network configuration

The first step is to check the network configuration of both endpoints and the intermediate devices, such as routers, switches, firewalls, and gateways. You need to verify that the IP addresses, subnet masks, gateways, and DNS servers are correct and consistent. You also need to ensure that the network devices support the protocols and codecs used by the VoIP system, such as SIP, RTP, G.711, or G.729. Additionally, you need to check the routing tables, access lists, NAT rules, and QoS policies to make sure that they allow the bidirectional flow of VoIP traffic.

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2 Test the connectivity and latency

The next step is to test the connectivity and latency between the endpoints and the VoIP server or provider. You can use tools such as ping, traceroute, or pathping to measure the round-trip time, packet loss, and hops of the VoIP packets. You can also use tools such as Wireshark or tcpdump to capture and analyze the VoIP packets and look for errors, delays, or drops. Ideally, you want to have a low latency, less than 150 ms, and no packet loss for a good VoIP quality. If you find any connectivity or latency issues, you need to troubleshoot the network devices or links that are causing them.

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Desmond Durrant

Sr. Lead Azure Cloud Architect/Developer | Cloud Security Strategy
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Network testing tools such as netperf can perform latency tests plus throughput tests and more. In netperf, the TCP_RR and UDP_RR (RR=request-response) tests report round-trip latency. With the -o flag, you can customize the output metrics to display the exact information you're interested in. Here’s an example of using the test-specific -o flag so netperf outputs several latency statistics: netperf -H <ip.address> -t TCP_RR -- \ -o min_latency,max_latency,mean_latency *Note: this uses global options: -H for remote-host and -t for test-name with a test-specific option -o for output-selectors.

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3 Check the audio settings and devices

The third step is to check the audio settings and devices of the endpoints. You need to make sure that the speakers, microphones, headsets, or phones are properly connected, configured, and working. You can use the sound test or echo test features of your VoIP system or software to verify the audio quality and volume. You can also check the audio settings of your operating system or application and adjust them if needed. For example, you can change the default audio device, mute or unmute the microphone, or disable any noise cancellation or enhancement features.

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Desmond Durrant

Sr. Lead Azure Cloud Architect/Developer | Cloud Security Strategy
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The third step in troubleshooting audio issues involves verifying the audio settings and devices on both endpoints. Here are the key actions to take: Check Audio Settings: Ensure that the microphone, speakers, and headphones (if applicable) are properly connected to the computer or device. Access the audio settings in the operating system or application being used. Verify that the correct input and output devices are selected. Adjust the volume levels to an appropriate level. Test Audio Devices: Use the built-in audio test or a third-party tool to verify that the microphone and speakers are functioning correctly. Test the audio playback by playing a sound or music file to check if you can hear it clearly. Test the microphone by recording.

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4 Check the firewall and NAT settings

The fourth step is to check the firewall and NAT settings of your network or device. Firewall and NAT are security features that can block or modify the VoIP traffic and cause one-way audio. You need to make sure that your firewall and NAT allow the VoIP ports and protocols, such as UDP 5060 for SIP and UDP 16384-32767 for RTP. You also need to enable SIP ALG (Application Layer Gateway) or SIP Fixup on your firewall or NAT device, which can help to handle the SIP signaling and RTP media streams. Alternatively, you can use a VPN (Virtual Private Network) or a STUN (Session Traversal Utilities for NAT) server to bypass the firewall or NAT issues.

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Christina Edenfield

Passionate DevOps Engineer | AWS Certified, Automation
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In my past experience with VoIP and troubleshooting one way audio, it was most of the time a firewall issue. Either the signaling of the VoIP call didn’t set up properly (an ACK not received to a 200 OK - the call would eventually timeout / drop) or a firewall blocking RTP.

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Alexander Lamazhapov

SaaS Support Engineer | Dedicated to Enhancing User Experience | Service Excellence Advocate | Ex-Zoom
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The most effective way to "prove" the issue on either side is to capture SIP and RTP data for the same call on both sides and then share packet captures. Need to identify whether the RTP packets are coming from the other end. In most cases, the customer issues with one way audio (especially if the customer onsite cannot hear far end), the culprit was a customer firewall or SBC. Sometimes double NAT can cause issues too.

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5 Check the bandwidth and QoS settings

The fifth step is to check the bandwidth and QoS (Quality of Service) settings of your network or device. Bandwidth and QoS are factors that can affect the VoIP performance and quality. You need to make sure that you have enough bandwidth to support the VoIP traffic, especially if you have other applications or users sharing the same network. You can use tools such as speedtest or iperf to measure your bandwidth and compare it with the VoIP requirements. You also need to enable and configure QoS on your network devices or endpoints, which can prioritize the VoIP traffic over other types of traffic and reduce the jitter, delay, and packet loss.

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6 Check the VoIP server or provider settings

The sixth and final step is to check the VoIP server or provider settings. You need to make sure that your VoIP server or provider is up and running and has the latest updates and patches. You also need to verify that your VoIP account, credentials, and registration are valid and active. You can use tools such as SIPp or sipsak to test the SIP registration and call scenarios. You can also contact your VoIP server or provider support and report any issues or errors that you encounter. They may be able to help you troubleshoot and resolve the one-way audio problem.

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7 Here’s what else to consider

This is a space to share examples, stories, or insights that don’t fit into any of the previous sections. What else would you like to add?

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Leonardo Quintero

PEng. | DevOps | Network Automation | VoIP | Always looking to work smarter, not harder
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Remember, one source of potential issues with audio, including mute calls , is the transcoding. Also, check connection information, maybe RTP is not going to the right interface.

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Abin Mathew, P.Eng

Carrier Voice Networks, VoIP, Contact Center | Ribbon, Cisco, Metaswitch, Nokia | SIP, RTP, SS7, H.248 | Solution Design | Advanced Troubleshooter | Implementation
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The common root causes: - Missing routes - Firewall blocking media ports - Codec issues. Analyze the sip trace for errors, codec negotiation, direction of media (send receive) & media ports If problem is still not identified, take a pcap trace from both the sending and receiving device interface. Use Wireshark to verify that the RTP stream is seen in both direction. Recently when analyzing a pcap trace we saw 2 way RTP stream but noticed the sending device switched the destination media ports after 2 packets. There was no signaling request to change the ports so this turned out to be a defect.

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Ahmed Alkholy

VoIP | Network | IT Engineer
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This in common cases problem in Nat traversal Check Stun server on wan And check sip Ports on wan And by checking RTP ports

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How can you troubleshoot one-way audio in a VoIP call?

1

2

3

4

5

6

7

1 Check the network configuration

2 Test the connectivity and latency

3 Check the audio settings and devices

4 Check the firewall and NAT settings

5 Check the bandwidth and QoS settings

6 Check the VoIP server or provider settings

7 Here’s what else to consider

Network Engineering

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How can you troubleshoot one-way audio in a VoIP call?

1

2

3

4

5

6

7

1 Check the network configuration

2 Test the connectivity and latency

3 Check the audio settings and devices

4 Check the firewall and NAT settings

5 Check the bandwidth and QoS settings

6 Check the VoIP server or provider settings

7 Here’s what else to consider

Network Engineering

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