I have a issue in my office hospital network regarding using IPPBX software.

[jobsp90]

We have a dedicated call center in our hospital. We are using a IP PBX software from a outsourced vendor. The software is installed in our PCs to attend the enquiry calls.The client pcs using in call center is a data vlan in one of the access switch. The server IP PBX software is installed in data center server farm switch. For the past weeks, the software was having some issues at some time, when i checked with the vendor they said software side is ok upon checking on wireshark logs. They have made several observations & fixes -

Key Observations:
Wrong Sequence Number (Red Entries) - RTP packets are arriving out of order or being lost, likely due to network congestion or QoS issues.
High Jitter Values (Max: 7.03 ms) - Increased jitter can degrade voice quality, causing choppy audio or lag.
Packet Loss (0.00%) - No significant packet loss was observed, but sequence errors suggest potential packet reordering.
Clock Drift (-79 ms) - A lack of synchronization between sender and receiver clocks can impact audio quality.

Cause - Solution.
Network Congestion - Prioritize RTP traffic using QoS settings on the firewall or router.
Jitter & Delay - Implement jitter buffers or increase buffer size in VoIP settings.
Packet Reordering - Check network routing and MTU settings to prevent out-of-order delivery.
Clock Drift - Ensure both endpoints synchronize time using NTP.

Pls help me what to do from the core switch side or on the access switch side.
NOTE- When I checked the ping status from the client PC to Server IP there is no drop or that much latency.
Core switches are CISCO 4506 model
Access switch for call center department is Aruba Instant ON 1930 model.

 

[HollTel]

Have you considered using my a Voice VLAN to keep your data and voice separate so that your QOS would be easier to track and manage. Your switch definitely has that ability, I would ask your software vendor if they do.

 

[DavetheChef]

It’s hard to advise without seeing the captures of the traffic for a bad call. Capture closest to the end that experiences the defects in its received voice.
A Windows PC cannot mark voice packets for QoS without a local or domain level policy. Your network may not trust these markings without some config work. Your network could also classify and mark the traffic if needed. Please don't even go to "my network has plenty of bandwidth and I don't need QoS".

For Jitter, pay attention to the max interpacket delay. This will tell you what your peak jitter is and not just a rounded average of the jitter. If the max delay between packets exceeds the packetization rate, you will hear some defects. Some PC applications do not have an automatic adaptive jitter buffer. You may be required to set it manually. Even for those applications that do have an adaptive buffer, keep in mind they are reactive, so you will still hear some defects, but less than a buffer which is set too low. most applications now also can smooth out a single packet loss by averaging the voice through the blank space, but this will also be noticed with a trained ear.

It says there is 0% packet loss but maybe you have out or order packets?
If there are out of order packets, then your network is likely sending packets along multiple physical paths 'packet by packet' rather than 'flow by flow'.
Clock Drift means so many things. NTP does not have a lot to do with voice, but it can affect the way the encryption operates if you are trying to secure the voice and signaling. Voice packets carry timestamps that tell the far end when to play them out and if there are long silent periods, depending the session setup.

If you want to understand how your voice is being received at the far end, look into the RTCP frames which are generally sent every 5 seconds throughout the call.