In today’s environment, organisations depend on real-time communication solutions like VoIP (Voice over Internet Protocol) to make remote work, sales calls, service centres, and internal collaboration possible. But it is still hard to make sure that voice calls are clear even when the network conditions change. Upcall Software solves this problem directly by using dynamic codec allocation and custom traffic filtering. This gives businesses call quality that is as good as what they would get with expensive infrastructure changes.
This article talks about how Upcall improves VoIP performance in real time, making sure that users can have clear conversations no matter what happens with bandwidth, packet loss, or network congestion.
Table of Contents
Learning the Basics
Before you learn about Upcall’s special optimisation methods, you need to know how the technologies work.
Dynamic Codec Allocation is the technique of automatically choosing and switching audio codecs during a call based on how the network is doing at the time. The system doesn’t just use a predetermined codec like G.711 or G.729. Instead, it looks at the current bandwidth, jitter, and latency and picks the codec that will give the best mix of quality and efficiency.
Traffic shaping is the process of controlling the flow of network data to give specific types of traffic, such as voice packets, priority. This keeps delays to a minimum and prevents congestion. It’s an important way to make sure that calls are always clear, especially when networks are shared with apps that use a lot of data.
Problems with the quality of enterprise VoIP
Network circumstances in a business might change a lot depending on where you are, what department you’re in, and what time of day it is. VoIP calls are especially susceptible to:
- Jitter: Voice distortions happen when packets arrive at different times.
- Latency: When packets take too long to get to their destination, it makes conversations unpleasant.
- Packet Loss: When data packets are missing, the audio sounds choppy or robotic.
Static codec selection makes these problems worse. For instance, utilising high-bitrate codecs like G.711 in a low-bandwidth setting can cause calls to drop or make them hard to understand. On the other hand, lower-bitrate codecs could compress audio too much when there is enough bandwidth, which lowers quality for no good reason.
Cloud apps, video conferencing tools, and large data transfers are also common uses for enterprise networks. If you don’t prioritise correctly, voice packets can be lost in the shuffle, which makes the experience worse for the user.
The Smart Dynamic Codec Allocation from Upcall
Upcall solves these problems by adding smart codec switching that happens in real time.
The software keeps an eye on network metrics, including packet loss rates, available bandwidth, and latency spikes, all the time. It dynamically assigns codecs like these based on what it learns:
- Opus is great for changing conditions because it can handle speeds from 6 to 510 kbps.
- G.711: High-quality audio that isn’t compressed and works on steady networks.
- G.729 and G.722 are compressed codecs that find a good mix between quality and low bandwidth utilisation.
Upcall’s switching method is what makes it stand out. If the network is too busy during a live connection, the system automatically switches to a more efficient codec like Opus or G.729 without stopping the call. This flexibility makes voices clearer and makes sure that calls don’t drop or get worse when there is a lot of pressure.
Upcall also uses heuristics and machine learning techniques to guess how traffic will change over time based on past data and how users act. This foresight lets you make codec changes before they happen instead of after they happen.
Custom Traffic Shaping Mechanisms
Codec switching isn’t enough on its own; voice packets still have to compete with other network traffic. This is where Upcall’s unique traffic shaping framework comes in.
The program uses QoS (Quality of Service) rules to give VoIP packets priority over other types of data. It finds VoIP traffic using protocols like SIP (Session Initiation Protocol) and RTP (Real-Time Transport Protocol), and then sends it along the best channels utilising:
- Tagging with DSCP (Differentiated Services Code Point) to give voice priority
- Reserving bandwidth for call sessions in situations with multiple tenants
- Dynamic queuing that changes the size of the buffer to cut down on jitter
Businesses can set up Upcall’s traffic shaping rules to work with the way they communicate with each other. For instance:
- Put calls with executives and customers ahead of internal chat audio.
- Send international calls over SIP trunk channels that have been optimised.
- Give VoIP traffic a higher priority during busy business hours.
In big companies, Upcall works with SD-WAN solutions to make sure that traffic rules are followed in offices that are spread out across a wide area. This makes sure that the quality of the audio remains the same from start to finish, even when users are connecting using public internet or mobile networks.
Benefits for Performance and Examples
The combination of dynamic codec allocation and traffic shaping leads to big improvements in performance. What Upcall users usually say is this:
- Call drop rates can go down by as much as 45%.
- 30–60% better MOS (Mean Opinion Score) for the quality of voice
- Real-time codec switching makes it easier to recover from jitter or packet loss faster.
- Fewer support tickets about VoIP problems
Case Study: Global BPO with More Than 5,000 Agents
An international BPO company with call centres in Asia, Europe, and North America used Upcall to replace its old VoIP management system. They had trouble with call quality, especially when the connection was poor in some areas.
After using Upcall:
- All time zones had clearer calls.
- The number of complaints about VoIP went down by 58% for the company.
- Codec efficiency saved them 20% of their bandwidth, which let them grow without having to build more infrastructure.
This change made it possible for the company to lower its operational costs and boost customer satisfaction at the same time.
Security and Compliance Considerations
Improving call quality should never put security at risk. Upcall makes sure that all codec discussions and traffic shaping happen over encrypted communication channels. TLS (Transport Layer Security) is used for signalling, and SRTP (Secure RTP) protects RTP streams.
Upcall’s codec and traffic management policies can also be completely audited, which helps businesses stay in line with standards like:
- GDPR stands for the General Data Protection Regulation.
- The Health Insurance Portability and Accountability Act (HIPAA)
- ISO 27001: Information Security Management Systems
Upcall also lets IT admins keep track of codec usage, spot problems, and make sure that all voice traffic stays under safe limits.
AI and other improvements in the future
Upcall is putting a lot of money into AI-driven speech enhancement for the future. Some of the new things that will be coming out soon are:
- Predictive Codec Scheduling: AI algorithms that predict traffic jams and switch codecs before they start to degrade.
- Self-Healing VoIP Routes: If a better-performing network path becomes available during a call, the system will immediately reroute the call.
- Voice Quality Analytics Dashboard: Get real-time information on how users are experiencing the network and how well the codec is working.
Upcall also wants to make its interaction with UCaaS (Unified Communications as a Service) platforms like Microsoft Teams, Zoom Phone, and RingCentral even better. This will make it easier to monitor VoIP quality across different ecosystems.
Conclusion
VoIP quality is becoming a competitive advantage as more and more firms rely on remote communication and working together with people all around the world. Upcall’s real-time optimisation engine, which uses dynamic codec allocation and unique traffic shaping, makes sure that businesses can send quality audio even when network circumstances are quite unexpected.
IT leaders can greatly increase voice dependability, customer engagement, and the future-proofing of their communication infrastructure by using these technologies. They can achieve all of this without spending a lot of money on new gear.
