Call Quality (no audio, 1-way audio, poor audio quality)

Common Call Quality Issues

Step-by-Step Troubleshooting

1. Determine Scope: Is it a single user, site-wide, or systemic issue?
   • Single User: Likely a local network issue.
   • Multiple Users, Same Site: Likely site-level network congestion or routing issue.
   • Multiple Sites or Random Users: This could be an upstream carrier, ISP, or host-side degradation.
2. Verify the user's environment by testing local network conditions.
   • Check the Network Type: Is the user on Wi-Fi or Ethernet? Ethernet is preferred.
   • Run a traceroute from the affected site to identify routing delays.
     • Look for:
       • Intermittent packet drops.
       • Long hops or geographic misrouting.
       • Latency (Ping): Ideal <150ms round trip.
     • Reboot Network Hardware: A simple power cycle of your router, firewall, or other network hardware can resolve audio issues by clearing temporary glitches and refreshing the configuration. 

ping -n 100 [15.222.191.147]
tracert [15.222.191.147]

3. Check Firewall / NAT / ISP Issues.
   • SIP ALG: Disable SIP ALG on the router/firewall.
     • SIP ALG often interferes with call setup and RTP stability.
     • Check the router/firewall admin page or CLI.
   • Port Blocking: Ensure the firewall allows UDP ports typically used for RTP (e.g., 10000–20000 ) and SIP (5060)
   • Enable QoS for Hosted Voice: Prioritize UDP ports RTP (e.g., 10000–20000 ) and SIP (5060).
   • Double NAT: Avoid multiple NAT devices between the endpoint and the internet. Ensure only one router is performing NAT.

4. Review device configuration: If the issue is isolated to one user or device, it's likely a local problem.
   • Confirm SIP Registration: Navigate to Users> Devices> check registration status in the portal.
   • Isolate the Device: Test from a different device or extension on the same network.
   • Firmware: Ensure phones, ATAs, or softphones are on the latest supported firmware.
5. Analyze the call 
   • Review and record recent call examples (within 3 days) through call history.
     • Type of issue experienced
     • Record the "called from number.”
     • Record the "called number.”
     • Record the "time and date.”
     • Record the "MOS score". Packet loss above 1% or jitter exceeding 30ms will noticeably impact voice quality.
       • Look for:
         • Packet Loss: Should be <1%.
         • Jitter: Should be <30ms.
     • Record the "call trace" or "PCAP" if possible.
       • RTP Stream stats:  Look for high jitter or packet loss.
       • Call Legs: Compare inbound vs. outbound audio quality.
       • INVITE/200 OK Delay: May show excessive setup time.
       • Call Disconnect Cause Codes.

Cradle to Grave: In the portal, navigate to Call History and filter the results to find the calls with quality issues. Select the "Cradle-to-Grave" icon to view the user-friendly call flow. 

The Cradle to Grave screen opens.

SIP Flow and Link: In the portal, navigate to Call History and filter the results to find the calls with quality issues. Select the "SIP Flow" icon to view the SIP flow detail. 

The SIP Flow screen opens. Click the "Share" button to copy the link to your clipboard. If you escalate the issue, retain this link to share with the support team.

What to look for in the Call Trace

Call Legs

Each call shows multiple legs:

• Originating Leg (A leg): Caller
• Terminating Leg (B leg): Destination
• Sometimes a Tandem Leg (e.g., forwarding or ring group)

Compare the signalling and RTP flow across these legs to pinpoint issues.

 RTP Stream Details are available in a tool like Wireshark

• Codec used (e.g., G.711, G.729)
• Packet Loss
• Jitter
• Media IP addresses and ports

RTP between internal legs should match. Discrepancies often point to firewall/NAT issues or wrong SDP handling.

SIP uses status codes to describe the outcome of requests (similar to HTTP). When a call fails, the response code from the far-end SIP entity (e.g., carrier, device, or system) will typically be in the 4xx or 5xx range.

6. Escalation Checklist

Before escalating a call quality or failure issue, gather:

• Record the extensions or numbers affected.
• Record the timestamp and time zone.
• Record symptoms and duration.
• Call Trace link.
• Record any relevant IPs or device models.

Hearing Radio Stations or Other Conversations During a Call

What it sounds like: A user hears faint music, a radio broadcast, or fragments of someone else's conversation bleeding into the background of their call. It often comes and goes, and may be easy to dismiss at first as background noise in the office.

Why it happens: Think of it like picking up a neighbouring radio station on an old AM car radio when you drive under a power line. Certain conditions allow an outside signal to sneak into the audio path. The cause and the fix depend on how the user's phone is connected to the Hosted PBX system, so the first step is always to identify the setup type.

Scenario 1: Native SIP IP Phone or UC Advanced Softphone

A SIP phone communicates over your data network using digital packets, just like your computer sends an email. There is no analog signal involved, so traditional telephone cable issues like radio pickup on copper wire do not apply.

In this scenario, radio crosstalk or interference is almost always caused by factors in the device's immediate physical environment rather than the Hosted PBX platform itself.

Common causes:

Wireless device interference is the most frequent culprit. Cordless phones, DECT handsets, baby monitors, two-way radios, and even some Bluetooth devices can broadcast on frequencies close enough to cause audible interference, particularly when they are placed near the IP phone or the workstation running the UC Advanced Softphone.

Electromagnetic interference (EMI) from electronics placed too close to the phone or its Ethernet cable can also introduce noise into the audio path. Common offenders include monitors, power bars, uninterruptible power supplies, and wireless chargers.

Poor or damaged Ethernet cabling running near a high-voltage electrical conduit can pick up electrical noise and degrade audio quality, though this is more typically manifested as choppy audio rather than radio crosstalk.

Troubleshooting Steps for SIP Phones and Softphones

Step 1: Switch to the UC Advanced Softphone. If the user is on a desk phone, have them test a call using the UC Advanced Softphone app on their computer or mobile device. If crosstalk disappears, the issue lies with the physical desk phone or its immediate environment, not the Hosted PBX platform.

Step 2: Clear the area around the phone. Move any cordless handsets, DECT base stations, two-way radios, or wireless devices away from the IP phone. Even shifting a device a metre or two can make a noticeable difference.

Step 3: Check for EMI sources. Move the phone away from monitors, power bars, or other electronics. Try a different power outlet if the phone uses a power adapter.

Step 4: Inspect the Ethernet cable. Replace the cable connecting the phone to the network switch or wall jack, particularly if it is old, coiled tightly, or running alongside power cables or electrical conduit.

Step 5: Test on a different network port or switch. If possible, plug the phone into a different wall jack or switch port to rule out a faulty port.

Step 6: Escalate if the issue persists. If the user is on a fully digital setup and the problem continues after the steps above, contact UnitedCloud support. In rare cases, the issue may relate to a carrier trunk or platform configuration that requires investigation.

Scenario 2: Analog Desk Phone Connected via an ATA

Some users connect a traditional analog desk phone to the Hosted PBX through an Analog Telephone Adapter (ATA). The ATA acts as a bridge, converting the digital SIP signal into an analog signal for the last stretch between the adapter and the phone. This analog leg of the connection reintroduces the same vulnerabilities found in traditional phone lines, and is where radio crosstalk is most likely to occur in a Hosted PBX environment.

Common causes:

Older, untwisted telephone cable between the ATA and the phone or wall jack can act like an antenna, picking up radio frequency interference from nearby AM broadcast towers, strong two-way radio systems, or other RF sources. This type of cable, often identifiable by its red, green, yellow, and black conductors, was not designed for environments with multiple lines or high RF activity. Replacing it with standard twisted-pair telephone cable typically resolves the issue.

Cordless phones and wireless devices near the ATA or the analog phone can introduce interference on the analog leg, for the same reasons described in Scenario 1.

Inline devices and splitters, such as standalone caller ID boxes or line splitters connected between the ATA and the phone, can introduce noise and interference into the analog signal path.

The ATA placed near other electronics can pick up electromagnetic interference, which is converted into audible noise on the call.

Troubleshooting Steps for ATA Setups

Step 1: Isolate the analog leg. Connect a single corded analog phone directly to the ATA, removing any splitters, caller ID devices, or other inline hardware. If the crosstalk clears, one of those removed devices was the source.

Step 2: Check and replace the telephone cable. Look for older, untwisted cable running between the ATA and the phone or wall jack. Replace it with standard twisted-pair telephone cable.

Step 3: Move the ATA. Relocate the ATA away from routers, monitors, power bars, and other electronic devices to reduce electromagnetic interference.

Step 4: Check for nearby wireless devices. Move cordless phone base stations, baby monitors, or two-way radios away from the ATA and the analog phone.

Step 5: Test with the UC Advanced Softphone. Have the user place a test call using the UC Advanced Softphone. If the crosstalk is gone, the issue is confirmed to be in the ATA setup or the analog wiring, not the Hosted PBX platform.

Step 6: Escalate if needed. If the steps above do not resolve the issue, contact UnitedCloud support with details about the ATA model, cable type, and troubleshooting steps already completed.

It is worth setting expectations with users upfront. Radio crosstalk is almost always a physical environment issue, not a Hosted PBX problem. The cloud phone system delivers a clean digital signal to the device. What the user hears is shaped by the local hardware, cabling, and wireless environment. Resolving it usually means improving that local setup rather than changing anything in the platform.