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Driven by a competitive telecommunications carrier (CLEC) that provides low-cost telephony services, Voice over IP (VoIP) is ready for large-scale deployment and integrated into established telecom operators (ILEC) In the network.
However, the lack of quality of VoIP services has long hampered its true threat to the traditional old-fashioned telephone service (POTS). However, this situation will no longer exist in the future. Chipmakers are taking significant steps to ensure that voice quality of service (QoS) over IP networks is exactly the same as POTS.
VoIP brings new opportunities to service providers. Because VoIP integrates data and voice services, it can be used in a variety of applications. In order to meet the unique requirements of different applications, a variety of optimized chip solutions have emerged. This article explains how these advanced chipsets and system-on-a-chip (SoC) devices meet the strict QoS requirements of service providers, and explains how they can implement versatile, high-quality, fully integrated VoIP system designs—these systems will be VoIP technology. The overall success of the contribution.
QoS Performance Parameters and Function Modules One term often encountered in the VoIP world is QoS. For VoIP, its importance is mainly reflected in the voice quality. The human ear is very sensitive to delays, background noise, and other line interferences that are typically caused by echo, jitter, and packet loss in VoIP. In order to achieve high voice quality, all components of a VoIP solution must be tuned in an optimal way.
To better understand how new solutions overcome QoS challenges, let's first look at what these challenges are.
◠Jitter - refers to the irregularity of the arrival of voice packets. A typical speech source generates a voice packet at a constant speed. However, in IP networks, packets do not always arrive at their destination in the original order—in other words, the rate of arrival is not constant and therefore causes jitter.
◠Echo—refers to the echo of your own voice. In other words, an echo is a leak that occurs during the transmission of sound to the receiver.
◠Latency—also known as delay—refers to the time it takes for a voice signal to travel from the origin to the end point across the network. The human ear is very sensitive to delays of more than 50 ms. Since IP is a "best-effort" connection, voice packets in the data network often experience delays.
◠Silent phase – refers to the stage in which the party on the call listens to the other party. In this case, it is not necessary to send a silent packet.
◠Packet loss/late packet/early packet—These packets cannot be transmitted to the voice stream in time, so the voice information is discarded. The higher the number of such packets, the lower the voice quality. Bad Frame Masking (BFM) - also known as Bad Frame Migration (BFI) or Lost Packet Concealment (PLC) - can cover some corrupted information.
VoIP system components
Figure 1 shows the main components of a VoIP system.
Figure 1 VoIP system components
Figure 2 Typical VoIP package
In a VoIP system, voice data is received (downstream) in the form of an IP packet. In the network processor, packets with the correct IP address (labeled "Voice Service") are selected and the IP header is removed. The Generic Datagram Protocol (UDP) header then determines the correct voice port. Figure 2 shows a typical VoIP package. After removing the UDP header, the RTP (Real Time Protocol) header is sent to the jitter buffer.
Jitter buffering has a very important impact on the voice quality of VoIP networks and devices. The task of the jitter buffer is to store voice packets to cover packet jitter. In addition, IP networks do not have a fixed transmission path, so each packet can choose a different route from the start point to the destination. This means that packets rarely arrive in the same order as they were sent. The jitter buffer rearranges the packets and determines the correct order by determining the arrival time and adapting to changes in network time. VoIP systems typically employ an adaptive jitter buffer that best accommodates the dynamic nature of IP networks. Finally, as mentioned above, the human ear is very sensitive to speech quality. Therefore, the jitter cache must be optimized to minimize jitter and latency without causing buffer under-runs, resulting in voice interruptions.
The jitter buffer works closely with the playback unit. The playback unit is responsible for playing the appropriate data packets at the correct time. If there are too many packets in the jitter buffer due to the higher sampling rate at the remote site, the playback unit must drop the packet or sample. If there is no packet, it must compensate for the speech interval with similar data.
In the parallel direction, the speech event and the silent phase must be probed and the appropriate pitch or noise needs to be generated.
After the playback unit, the speech is decompressed and then sent to the digital to analog converter, which in turn is transmitted to the Subscriber Line Interface Circuit (SLIC). The SLIC performs a 4-2 line conversion (mixing) and then sends the signal to the analog phone.
At the same time, in the upstream direction, or referred to as the transmission direction, the analog voice signal from the telephone is sent to the voice processing unit via the SLIC and the digital to analog converter. The first device here is the Line Echo Cancellation (LEC) unit. Due to a mismatch between the rated impedance of the client device or line card and the actual impedance of the phone, the transition to the 2-wire interface causes an echo. The echo trailing pulse length of an analog phone is typically in the range of 4ms. In an IP environment, all VoIP providers must ensure that system echoes are minimized. Since latency is usually higher in IP systems, it is critical to achieve echo cancellation. The best results are obtained when the echo is removed to near the original state. This is called near-end echo cancellation.
Once the echo is removed from the system, the speech is compressed and encapsulated into RTP packets. At the same time, signaling events, such as modem tones or DTMF (Dual Tone Multi-Frequency) tones, are detected and signaled to the network processor, automatically converted to event packets by the encoder if necessary. The voice packets are then transmitted to the processor where their corresponding UDP and IP headers are added and then sent to the Ethernet port.
The process described above is only relevant for voice calls. Fax calls use T.38 fax relay for different processing to improve transmission quality and performance. T.38 is a protocol running on a network processor that is responsible for the "connections" of both parties. In addition, T.38 requires that the fax signal be modulated and demodulated into packets. The fax call probe is the same as the tone probe described above, and instead of the compression/decompression algorithm is the modulation/demodulation firmware.
To achieve high quality of service, VoIP requires good interaction between different components of a VoIP device. The transmission of voice packets must be done in the shortest possible time, and all packets must be processed as quickly as possible to ensure the same voice quality as traditional telephone services. Infineon Technologies' VINETIC voice processor family overcomes these challenges. By integrating codecs and voice processing into a single device, VINETIC is able to package all real-time and performance-critical functions together. This makes it easy to implement a modular configuration with multiple voice ports as needed, using the same network processor and software.
VINETIC Voice Processor Infineon Technologies' VINETIC voice processor family includes a number of flexible devices that are pin and software compatible and available in 2 analog ports and 4 analog port versions. For VoIP, VINETIC-VIP and VINETIC-M are dedicated devices that integrate package functionality into the device. The VIP version has full functionality: G.729A/B/E, G.728, G.723.1, G.726 and G.711 compression and T.38 fax relay data pumps. The M version is slightly smaller, allowing the voice to take up more bandwidth by using G.711 or G.726, resulting in lower prices.
The VINETIC family also includes a variety of SLIC devices optimized for the specific requirements of different applications, such as the cost-optimized CPE ringing SLIC and the CO-level non-ringing and ringing SLIC.
Applications and solutions
After learning about the VoIP device process, let's look at the different VoIP applications. The following are some of the mainstream applications of VoIP, and different vendors have introduced a number of different applications.
1 Voice-enabled broadband router The first application is a broadband router with VoIP capabilities. Figure 3 shows a set of routers that provide an Ethernet uplink and multiple Ethernet downlinks. Wireless LAN (WLAN) is supported through the PCI interface of the network processor. The VoIP voice connection is added by using VINETIC-2CPE with a ringing SLIC connected to integrate voice directly into the data system. The FXO interface (relay interface) as a special option is also shown on the diagram, allowing local calls to be routed directly to the PSTN line.
Figure 3 Voice-enabled broadband router
Figure 4 analog phone adapter
2 Analog Phone Adapter The second application is the well-known analog telephone adapter (ATA), and Figure 4 shows the cost-optimized version. The same application in the field of cable modems is also known as the Independent Media Terminal Adapter (SMTA).
ATA is a voice-only application with no data services. It connects to a modem or router via an Ethernet interface. The second Ethernet port can be selected, but the data service is only converted. VoIP voice services are implemented through VINETIC-2CPE and cost-effective network processors or microcontrollers. This design enables system vendors to develop small VoIP devices for cost-sensitive market applications.
3 The third example of a VoIP line card is a VoIP line card that shows the characteristics of a multi-channel VoIP solution, as shown in Figure 5.
Figure 5 VoIP line card
The line card in Figure 5 reflects the modular features achieved by utilizing the VINETIC device. All POTS ports are in parallel and connected to a cost-effective network processor. Since voice processing and packetization are performed within the VINETIC device, the function of the processor is to act as a summary circuit for the different voice ports, transferring them to a single Ethernet interface.
The VINETIC-4VIP device is suitable for mature VoIP services including G.72x vocoders and T.38 fax relays. The VINETIC-4M and VINETIC-4S are two cost- and cost-compatible devices that are compatible with each other and are slightly less expensive than the VINETIC-4VIP.
The VINETIC-4M is a simplified version of the VINETIC-4VIP that provides only G.711 and G.726 encoding while maintaining a non-blocking VoIP system. VINETIC-4S uses the "shared line card" architecture. The VINETIC-4VIP device is mixed with the VINETIC-4S device (pure TDM device) on the line card while maintaining pin and software compatibility. Due to the connection to the VINETIC-4VIP device, the voice channel connected to the TDM device can be routed to the VoIP block on the 4VIP device. The result is a cost savings for line cards that take full account of voice port usage (Erlang coefficient) while keeping system vendors fully flexible.
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