--- title: Troubleshooting Voice Related Concerns (Disturbance & Breakage) slug: contact-center/troubleshooting-voice-related-concerns-disturbance-and-breakage docTags: createdAt: 2026-07-08T10:37:51.967Z --- **** ## Overview of Call Architecture A standard call is a combination of two distinct connections, commonly referred to as legs: - **Agent's Leg:** The connection between the agent's system and the call server. - **Customer's Leg:** The connection between the SIP line and the call server. Voice-related concerns typically arise due to environmental/system issues or network-related problems. ## Network Prerequisites For the agent's leg to successfully connect, specific ports must be open on the network. Ensure the following network configuration is applied: | Port | Protocol
  | | ----------- | --------------- | | 5060 | TCP | | 8089 | TCP | | 10000-20000 | UDP | ## Agent Connection Methods Agents' legs can be maintained using one of the following four options. Each method has specific configuration and troubleshooting requirements: ### 1. Softphone In a softphone setup, extensions are created on the platform (e.g., Ameyo) and registered on a third-party softphone application like X-Lite or Zoiper. Agents log in to the UI via a browser. - **Dependency:** Network connectivity between the agent's system and the call server is required for the extension to register. - **Risk:** If unregistered, calls will drop, and the agent will be placed in an erroneous state. - **Hardware:** Requires an external headset for audio transmission. ### 2. Hardphone / IP Phone Extensions are created on the platform and registered directly on the physical IP phone. - **Role:** These agents operate under a "phone agent" role. - **UI Access:** They typically do not use the browser UI and log directly into the hardphone/IP phone. ### 3. WebRTC The agent's leg is maintained directly on the browser, and extensions are auto-created. - **Dependency:** No reliance on third-party softphones or hardphone registries. - **Prerequisites:** Specific network ports (see above) must be open. The microphone must be successfully detected by the agent's system and browser, otherwise, WebRTC will not work. ### 4. Mobile Phone (Dual Leg) The agent selects the same SIP trunk used for inbound/outbound routing and provides their mobile number to route the call to it. - **Resource Usage:** For every single call, **two channels** of the SIP trunk are consumed (one for the agent's leg, one for the customer's leg). ## Initial Troubleshooting Steps for Voice Issues If an agent experiences voice issues, follow these initial diagnostic steps: 1. **Review Reports:** Download the call reports and verify if there is talk time for the reported call. 2. **Listen to Voice Logs:** Review the recordings of the reported call to determine if the issue originates from the agent's side or the customer's side. 3. **Cross-Check Customer Issues:** If the issue seems to be from the customer's side, listen to a few other voice logs from that customer/line to confirm. 4. **Verify Widespread Agent Issues:** If the voice issue is consistently on the agent's side across all calls, proceed with specific hardware/network diagnostics. 5. **Ping Test:** Ping the call server from the agent's system and check for any Request Time Out (RTO) or packet loss. 6. **TCP Dump:** Capture a TCP dump on both the agent's system and the call server to rigorously verify network connectivity. ## Specific Voice-Related Concerns and Resolutions ### Headset Issues Agents may report choppy/no voice from the customer, the customer being unable to hear the agent, or static disturbance whether on a call or not. - **Resolution:** Replace the headset with a known working one from another agent to verify if the hardware is faulty. If the issue is resolved by replacing the headset, permanently issue a new one. ### Agent IP Phone Issues Issues include voice disturbance or the inability to hear audio on either end. - **Resolution Steps:** - Verify the IP phone's registration status. - Verify the configuration settings done on the IP phone. - Test by swapping the IP phone with a working unit. - Reconfigure the IP phone if necessary. ### Network-Related Concerns (WebRTC) Voice issues in WebRTC setups are often caused by latency, jitter, or port blockages. A **WebRTC Troubleshooter** and Network Debugger are provided on the agent's screen (dial pad) to identify these issues. **WebRTC Signal Indicators:** The signal color on the agent's screen changes based on latency thresholds (which are configurable in the database): - **Green:** Latency is below the configured "low" value. - **Amber:** Latency is between the "low" and "medium/high" values. - **Red:** Latency is above the configured "medium" value. **Resolution Steps for WebRTC Network Issues:** - **Red Signal / High Latency:** Enable continuous ping from the agent system to the call server to monitor network stability. - **ICE Connection Failure:** Ensure UDP ports 10000-20000 are open on the network. - **Erroneous State on Login (Red Signal):** Check the network specifically for TCP port 8089 blockages. - **No Audio (Both Sides):** Verify that UDP ports 10000-20000 are open **bi-directionally**. - **Immediate Call Disconnect:** Check the admin screen to ensure the agent is mapped to the correct call server. - **Advanced Debugging:** In all the above cases, enabling a TCP dump on both the call server and the agent's system is highly recommended to isolate the bottleneck.