Voice over Internet Protocol (VoIP) technology, more so asterisk setting has significantly transformed communication, allowing voice data to be transmitted over the internet instead of traditional phone networks. For service providers and businesses using Asterisk, a popular open-source telephony platform, ensuring high-quality VoIP termination is critical. VoIP termination refers to connecting VoIP calls to the Public Switched Telephone Network (PSTN) or routing calls between different VoIP networks. Achieving superior call quality and reducing issues like latency, jitter, packet loss, and echo require proper configuration and optimization within Asterisk.
This article outlines the essential settings and best practices within Asterisk that contribute to high-quality VoIP termination.
1. Understanding VoIP Termination in Asterisk Setting
VoIP termination allows users to place and receive calls between VoIP networks and traditional phone systems, such as PSTN or mobile phones. Asterisk acts as a gateway, handling call routing, protocol conversions, voicemail, call queuing, and other telephony features while maintaining seamless connectivity with different networks.
To achieve the best VoIP termination quality, Asterisk must be configured correctly to handle voice traffic, minimize packet loss, reduce latency, and ensure good call clarity.
2. Network and Hardware Considerations
High-quality VoIP termination begins with a stable and well-optimized network and hardware infrastructure. VoIP calls are sensitive to network issues, so ensuring the right network setup is fundamental.
Network Configuration
· Bandwidth and Latency: VoIP calls require a stable internet connection with sufficient bandwidth. Insufficient bandwidth can lead to voice quality degradation, including dropped calls and choppy audio. Ideally, each active call should have between 80-100 Kbps of bandwidth, depending on the codec used. Low latency (below 150ms) is crucial for clear communication. Latency greater than this can cause noticeable delays in conversation, leading to poor user experience.
· Packet Loss: VoIP traffic should be prioritized on the network, ensuring minimal packet loss. Packet loss can lead to missing words, choppy audio, or call drops. Ensuring a low packet loss rate and optimizing the network for VoIP traffic helps maintain call integrity.
· Jitter Management: Jitter is the variation in packet arrival times, which can disrupt call quality. Asterisk provides jitter buffers, which temporarily store incoming voice packets to smooth out these variations. Fine-tuning these buffers ensures that voice calls are continuous and without distortion.
Hardware Setup
· Dedicated Servers: Using dedicated servers for Asterisk ensures that your system has enough CPU, RAM, and network resources to handle multiple concurrent calls without performance degradation. VoIP calls require considerable computational resources, particularly when transcoding between codecs or handling multiple calls.
· Digital Signal Processing (DSP): For higher-quality audio, especially in large-scale implementations, DSP hardware can help offload tasks such as transcoding and echo cancellation. This allows the Asterisk server to focus on routing and signaling.
3. Codec Selection
The codec used in VoIP calls directly affects both call quality and bandwidth consumption. Asterisk supports a variety of codecs, and selecting the most appropriate codec for your network and hardware is essential for optimal VoIP termination.
· G.711 Codec: This codec is one of the most widely used for VoIP, providing high-quality voice transmission. It uses uncompressed audio, offering excellent clarity and low delay, but requires more bandwidth (approximately 64 Kbps per call). This codec is ideal for high-quality calls where bandwidth is not a constraint.
· G.729 Codec: In contrast, G.729 is a compressed codec that reduces bandwidth usage to approximately 8 Kbps per call. While it consumes less bandwidth, it may compromise audio quality due to the compression. This codec is typically used in scenarios with bandwidth limitations or for international calls where cost-saving is essential.
· Opus Codec: Opus is an advanced codec known for its high efficiency and ability to adapt to varying network conditions. It provides better voice quality than G.729 at lower bitrates and is widely used in modern communication systems. Opus also supports both narrowband and wideband audio, making it versatile for various types of VoIP calls.
Choosing the right codec based on your network’s bandwidth and the required audio quality is crucial. Asterisk allows you to prioritize certain codecs over others to ensure that calls use the best codec available.
4. Jitter Buffer Configuration
Jitter is a common issue in VoIP communications, and Asterisk provides jitter buffers to handle variations in packet arrival times. A jitter buffer temporarily stores packets to ensure smooth audio playback by compensating for the variation in transmission times.
The size of the jitter buffer must be configured appropriately to handle the expected jitter without introducing excessive delay. A larger jitter buffer can handle more variation but may introduce latency if the buffer is too large. Proper configuration of the jitter buffer can significantly improve the call experience, especially in networks with unstable connections or high jitter.
5. Echo Cancellation
Echo is another common problem in VoIP communication, especially when dealing with PSTN networks. Echo can be caused by signal reflection or improper microphone or speaker placement. Asterisk includes built-in echo cancellation features to reduce or eliminate echo during calls, enhancing audio clarity.
Echo cancellation works by identifying and removing delayed voice signals that are returned to the caller’s side. Asterisk provides different levels of echo cancellation, which can be fine-tuned based on the environment and the hardware in use. Enabling echo cancellation improves the overall voice quality, especially in long-distance or international calls.
6. Silence Suppression and Comfort Noise
Silence suppression is an asterisk setting feature that prevents the transmission of silence during calls, saving bandwidth and improving call efficiency. However, calls may sound unnatural if silence is completely removed. To address this, Asterisk provides comfort noise generation, which injects synthetic noise during periods of silence, ensuring that the call feels continuous and natural.
By enabling silence suppression and comfort noise, Asterisk optimizes bandwidth usage while maintaining an uninterrupted call experience. This setting is especially useful in scenarios where bandwidth is a limiting factor.
7. Quality of Service (QoS) and Traffic Prioritization
For high-quality VoIP termination, ensuring that VoIP traffic is prioritized on the network is essential. Quality of Service (QoS) mechanisms are used to prioritize VoIP packets over other types of network traffic. By marking VoIP packets with higher priority, QoS ensures that voice traffic receives the necessary resources, even during periods of network congestion.
Implementing QoS on network routers and switches, and configuring appropriate settings in Asterisk, can significantly improve VoIP quality by ensuring that voice packets are transmitted with minimal delay and minimal risk of congestion or packet loss.
8. Call Routing and Load Balancing
In a large-scale VoIP system, effective call routing and load balancing are key to ensuring quality and reliability. Asterisk offers flexible routing options, allowing administrators to define specific routes based on the destination, time of day, call volume, or other criteria.
For high-volume environments, load balancing across multiple servers or trunks can distribute traffic evenly, preventing any single server or connection from becoming overloaded. Proper call routing helps ensure that calls are completed efficiently and that the highest-quality routes are chosen, reducing the chances of encountering call drops or quality degradation due to network congestion.
9. Monitoring and Logging
Continuous monitoring of Asterisk’s performance and VoIP call quality is essential for maintaining high-quality VoIP termination. By tracking metrics such as jitter, latency, packet loss, and call duration, you can identify and resolve potential issues before they affect the service.
Asterisk provides built-in logging and diagnostic tools that allow administrators to monitor call quality in real time. Using these tools, administrators can gain insights into network performance and VoIP quality, helping to troubleshoot problems such as dropped calls, poor audio quality, or call failures.
10. Security Settings
While security does not directly affect call quality, protecting the Asterisk system from unauthorized access and attacks is crucial for maintaining uninterrupted service. Ensuring that Asterisk is secured through measures such as firewalls, IP whitelisting, and secure transport protocols (like TLS and SRTP for encrypted signaling and media) helps avoid disruptions that could impact call quality. Ensuring that your system is secure helps maintain its reliability, which in turn helps preserve the quality of service for your VoIP termination.
Conclusion
Achieving high-quality VoIP termination with Asterisk requires careful configuration of both network settings and the platform itself. Prioritizing codecs, managing jitter, implementing echo cancellation, optimizing bandwidth, and configuring QoS all contribute to ensuring that VoIP calls are clear, reliable, and cost-effective. Additionally, regular monitoring and the use of security best practices ensure that the system remains stable and continues to deliver optimal performance over time. By understanding and optimizing these essential Asterisk settings, you can create a high-quality VoIP termination solution that meets the needs of businesses and service providers alike.
