Enterprise VoIP deployment: How to Ensure QoS

At present, many enterprises have two independent IP networks and telephone networks. A Wide Area Network of an enterprise's computer network system usually uses a router to connect an enterprise's LAN to an enterprise's IP network. The telephone network is composed of PBX or group phone numbers of each branch. In this way, the telephone calls between branches within an enterprise need to be implemented through the public long-distance telephone service. The cost of long-distance telephone has become a major expense of the enterprise's business activities. For example, a teleconference within an enterprise can last for several hours. Therefore, you can lease DDN, Frame Relay, and ATM leased lines to transfer enterprise IP networks and telephone networks over one link, which can greatly save the company's telephone overhead. (1)

: Implementation of long-distance VoIP

Use the voice gateway to convert voice information into IP data packets, which are transmitted over an IP network, that is, VoIP. Because the TCP/IP network is open and interconnected on a link, it has excellent performance and price ratio and is widely used. Currently, a large number of enterprise WAN routers have the voice gateway function.

Because TCP/IP-based data communication is essentially different from telephone speech communication. Voice communication is connection-oriented and sensitive to network bandwidth, latency, error rate, and jitter. The transmission of IP data packets is non-connection oriented. The communication attribute is abrupt, and the traffic changes. The traffic control is implemented by the End device. The jitter is also introduced to the IP data packets with a longer length. Therefore, implementing voice communication over an IP network requires the IP router to provide comprehensive QoS Assurance and call monitoring.

Taking the network as an example, this paper introduces one of the important steps to implement enterprise VoIP: QoS implementation.

QoS implementation

The QoS parameters of VoIP are as follows:

1. packet loss rate <1%.

2. One-way transmission latency <ms ~ 200 ms.

3. jitter <30 ms.

4. Each call requires 21Kbps ~ Bandwidth of 106Kbps (bandwidth requirements vary depending on the encoding method and link layer encapsulation ).

The loss of voice data packets may cause intermittent voice. Currently, the standard DSP chip can tolerate data loss within 30 ms. The packet loss rate is calculated at a sampling rate of 50 PPS per second. The packet loss rate must be less than 1%. If the packet loss rate exceeds 1%, the voice quality is unacceptable.

According to the ITU Standard, the one-way latency of high-quality voice must be less than 150 ms. When the one-way voice latency exceeds ms, people will feel the opposite side is too slow to reflect, just like talking over a walkie talkie. Based on practical experience, the end-to-end unidirectional latency of voice must be less than 200 ms.

Generally, VoIP devices provide a buffer to control Jitter to suppress jitter. The tolerable range is up to 30 ms.

Bandwidth requirements vary depending on the encoding method and link layer encapsulation. Generally, G.729A and G.711 are used as the encoding methods. HDLC is used as the link encapsulation, and the bandwidth requirements are 26Kbps and 81 Kbps respectively. When a PPP or frame relay link is used, the bandwidth is 28 Kbps and 84 Kbps. The number of ATMs is 43Kbps and 106 Kbps. Multi-link PPP is 30 Kbps and 86 Kbps.

To meet the preceding QoS parameters, consider the following two aspects when designing an enterprise's VoIP network:

1. Wan Link Selection

2. QoS implementation of The Voice Gateway Router

Considering the delay, jitter, and efficiency of the link, the WAN link should be the optimal leased line based on the services and prices provided by domestic operators. Frame Relay or ATM can be considered in some urban areas. The worst line error rate must be less than 10-5, otherwise the voice quality will not meet the requirements. The transmission delay of the link should be considered based on distance; domestic long distance should be controlled at One Direction 20 ms/1000 km, and international long distance should be controlled at one direction within ms. To ensure the stable operation of VoIP and other network applications, we recommend that you set the bandwidth to no more than 50% of the link bandwidth.

The Voice Gateway Router should have a sound QoS function. If the WAN bandwidth is less than 2 Mbps, to avoid Jitter Caused by long data packets, you should reduce the MTU of the link and set the MTU value based on the link bandwidth. When splitting IP packets, the router should not increase the processing latency too much. If an ATM or frame relay link is used, the router should support the QoS function that works with ATM or frame relay, such as port control signaling and traffic control.

The Voice Gateway Router should also have good port queuing and queue scheduling functions. An independent queue should be allocated for voice data packets and forwarding should be given priority. At present, from the perspective of practice, the hybrid queuing method of low-latency queues LLQ and CBWFQ of Cisco routers is the most effective. The principle is as follows:

Among them, LLQ ensures that voice packets can be forwarded first. When the link is congested, it ensures the minimum network latency and packet loss rate. CBWFQ ensures the stable operation of other applications. Fragment and Interleave can avoid network jitter.

: CBWFQ Algorithm

With QoS Assurance, high-quality voice can be provided under normal conditions. However, when exceptions occur, such as Link error rate increase, router busy, and route fluctuation, the speech quality will also decline. In this case, how to monitor the network status and provide the transfer of call routes, connecting the phone through the public phone network is an indispensable function of enterprise VoIP.

1. Increasing the cost of VoIP deployment remains a major obstacle
2. Effective deployment of VoIP
3. How the IT Department prepares for VoIP deployment