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.

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.

Best Practices to Avoid Future Issues

• Use business-grade routers with Cloud PBX QoS.
• Ensure wired Ethernet for critical endpoints.
• Educate clients to avoid congesting bandwidth (e.g., streaming, file sharing).
• Implement dedicated VLANs for voice traffic.
• Regularly review firmware and device health.