VoIP services and the Access Network

Access network support the VoIP services

  VoIP is the technology that the analog-to-speech signal is sampled, digitized, compressed and encapsulated into data packets, and then transmitted and processed in an IP network in the form of packet pairs.
  The voice service is a basic telecommunication service, and it is the most frequently used by people, and is also the service with the highest revenue among operators. Although relying on the existing TDM technology and network, it can well support the existing voice service, but it is developing towards IP throughout the entire network. Under the general trend, the adjustment of voice services to IP bearers needs to be gradually implemented to adapt to changes in the core network. After the traditional network facilities are eliminated, the ability to provide voice services is reduced.

VoIP gateway with 4G cellular data sim card

  Voice IPization is an important part network and ap[plication services IPization. Providing voice services over the Internet and IP-based networks can reduce service deployment and usage costs. VoIP is a very low-cost, universal, and easy-to-use communication method. The IP of voice will make voice services become a kind of business application on the Internet.

  For all operators, the development of the VoIP market is an important development basis, especially for operators that traditionally provide TDM voice services. It is not as good as looking at the business income being eroded by the VoIP of various ISPs on the Internet. Utilizing its own position in the business field also directly develops VoIP and strives to become the leader in the VoIP field.
  With the deepening of IP network access to the access layer, the access layer network has gradually become a network with IP technology as the core. It is used as the voice service that the access layer needs to support first, and the voice IP is implemented at the access layer. It has also become an inevitable requirement for the development of the access layer network.
  In the access layer network, voice VoIP needs to do the following work:

Voice Packetization

  The packetization of voice is accomplished through Code-Decoder (Codec). In order to reduce the bandwidth requirement, the voice needs to be compressed and encoded. Different compression and coding algorithms can obtain different rates of voice traffic. The more compression, the lower the voice quality, but the more processor resources are consumed. At present, several common Codec standards in the industry are described below.

  • G.711: It is currently the most widely used codec algorithm, which is equivalent to the current PSTN phone number flow rate. The sampling frequency is 8 kHz, and the bit rate is 64 kbps. The best audio quality is G.711.
  • G.723.1: The sampling frequency is 8 kHz, and the bit rate is 6.3 kbps and 5.3 kbps, respectively. Due to the low rate, the encoding side has a delay of 37.3 ms.
  • G.729: The sampling frequency is 8 kHz, and the bit rate is 8 kbps. Later, an enhanced version of G.729a was developed.

Voice service QoS guarantee

  VoIP is a kind of sensitive service to delay and delay jitter, and the end-to-end delay is less than 150ms. If this delay is exceeded, the two parties will feel a noticeable pause and the voice will be prolonged, affecting the quality of the call. Delay jitter has a great influence on the quality of VoIP. In order to remove jitter, a debounce buffer is generally used at the receiving end to absorb jitter.
VoIP delay consists of the following stages of delay.

  • Processing/packet delays: sampling, compression, decompression, packet and packet delays, and buffering delays in each queue;
  • Serialization delay: the delay of packets transmitted at the physical layer. The smaller the packet, the higher the line rate and the smaller the delay.
  • Transmission delay: time delay transmitted on the line facility;
  • Debounce delay: Debounced buffers also introduce delays. For this reason, smaller buffers are generally preferred.

  In order to guarantee the QoS of VoIP, it is necessary to pay attention to the following links.
Step1, VoIP flow tag
  VoIP flows are marked with a priority in their ToS field, carried in the entire domain, and are extended to all levels. All related layers are processed and distributed with higher priority.
Step2, Adopt queue technology that supports VoIP priority transmission

  • SP: Strict priority;
  • SP+WRR: Ensures that sensitive services are even scheduled, while low-level services can also be served;
  • IP RTP priority: Only RTP streams are prioritized and other services are WRR.

Step3, With echo cancellation technology, it can be built-in on Codec.

VoIP gateway with 4G cellular data sim card (FOT)

Provides a protocol interface to access the core network of IP voice services

  At present, the core network of voice services of the next-generation network is evolving to the next-generation switching network with softswitch/IMS as its core. The access layer network needs to provide a protocol interface to access the softswitch network/IMS network. The main protocols currently used are as follows.

1) MGCP
  MGCP(Media Gateway Control Protocol), Media Gateway Control Protocol. The gateway mentioned here refers to the gateway between the packet-switched network and the traditional circuit-switched network. The MGCP protocol includes the Simple Gateway Control Protocol (SGCP) and IP Device Control (IPDC) protocols. The SGCP protocol is a simple remote control protocol used to control voice gateways and network access servers. The IPDC protocol enables the public switched telephone network (PSTN) to seamlessly connect with the third layer of the IP network.

2) H.248
  H.248 is a gateway control protocol and is a formal standard issued by the IETF and ITU-T. As an MGCP evolution protocol, the H.248 protocol inherits many advantages of MGCP and has significant improvements over MGCP in terms of scalability, security, and interoperability. The H.248 protocol can quickly adapt to new business requirements and thus greatly advance H.248. It should be said that H.248 has replaced MGCP as the protocol standard between MGC and MG.

3) SIP
  SIP is a multimedia IP architecture proposed by the IETF. It is a protocol designed specifically for VoIP and IP multimedia services. SIP is a client-server protocol in text format: the client initiates a request and the server responds. SIP establishes, changes, and terminates calls between end-to-end users based on IP networks. For the telephone service, other standards and protocols are also needed, such as the RTP real-time transmission protocol. The SIP protocol has been rapidly developed due to its simplicity, ease of expansion, and ease of implementation. It is gradually becoming the basic protocol in Next Generation Network (NGN) and IMS.

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