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Voice Feature Guide 01

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Huawei uses machine translation combined with human proofreading to translate this document to different languages in order to help you better understand the content of this document. Note: Even the most advanced machine translation cannot match the quality of professional translators. Huawei shall not bear any responsibility for translation accuracy and it is recommended that you refer to the English document (a link for which has been provided).
Voice Media and Signaling

Voice Media and Signaling

Media and signaling play important roles in voice services.
  • Voice media: Used to carry and normally transmit voice communication contents. In the NGN system architecture, Real-Time Transport Protocol (RTP) carries media streams.
  • Voice signaling: Used to set up and control voice communication between two telecommunication entities. Different from IP protocols, signaling protocol fields carry commands. Common signaling protocols are MGCP, H.248, and SIP.

Voice Media

RTP

Real-Time Transport Protocol (RTP) is dedicated for multi-media streams over the Internet. In a VoIP network, RTP carries media streams. For details about RTP, see the RFC3550.

RTP provides end-to-end (E2E) services to transport real time data (including audios and videos) in the format defined in G.711, RFC2833, and RFC2198. Figure 1-7 shows the format of an RTP packet.
Figure 1-7 RTP packet format
RTP runs on top of User Datagram Protocol (UDP) to make use of its multiplexing and checksum services. However, RTP may be used with other suitable underlying network or transport protocols. Figure 1-8 shows the RTP protocol stack model.
Figure 1-8 RTP protocol stack model

RTP receives media streams from the upper-layer device and encapsulates the streams into RTP packets. Then RTP sends the packets to the lower-layer device. The lower-layer protocol transmits RTP and Real-Time Transport Control Protocol (RTCP) packets through different ports. For example, if UDP is used as the lower layer protocol, the protocol uses a port with the ID of an even number to transmit RTP packets and uses the port with the ID of the odd number following the even number to transmit RTCP packets.

RTCP

RTP itself ensures real-time data transmission, but cannot provide a mechanism for reliably transmitting data in sequence or a traffic and congestion control mechanism. It provides the mechanisms using RTCP.

RTCP is based on the periodic transmission of control packets to all participants in the session, using the same distribution mechanism as the data packets, so that it provides feedback on the quality of the data distribution.
  • RTCP packets take 5% of bandwidths.
  • RTCP packets contains ring delay, packet loss stream, and jitter for QoS monitoring.
Figure 1-9 shows the format of an RTCP packet.
Figure 1-9 RTCP packet format
RFC 2833

RFC2833 defines a dedicated format to reliably transmit important data, such as signal tone, event, dual tone multi-frequency (DTMF) signal in G.711 and G.729 communication.

Figure 1-10 shows the typical RFC2833 application scenario.
  1. After the user dial a number, the softswitch/IMS controls the access gateway (AG) to create and transmit RTP voice media streams to the media resource server (MRS).
  2. The MRS plays an announcement of dialing the number to the AG.
  3. The user dials the number. Then the number is transmitted to the MRS through the RFC2833 packet that is carried over voice media RTP.
  4. The MRS collects and transmits the number to the softswitch/IMS.
Figure 1-10 Typical RFC2833 application scenario
RFC2198 Redundancy
RFC2198 describes the RTP payload format for redundant audio data, which can be used for the RFC2833 digit collecting, fax transparent transmission service, and modem transparent transmission service.
NOTE:

RFC2198 redundancy is unnecessary for T.38 fax services, this is because T.38 fax service supports redundancy for its own.

RFC2198 improves the reliability of data transmission through redundant transmission. When the network quality is poor, redundant transmission can ensure the service quality and reduce impacts brought by distorted signals. Figure 1-11 shows RFC2198 redundancy application
Figure 1-11 RFC2198 redundancy application

Voice Signaling

The signaling technology implements phone calls. The commonly used VoIP control signaling systems contain MGCP, H.248, and SIP.

MGCP Protocol
Media Gateway Control Protocol (MGCP) is defined in the RFC2705 standard and it defines a call control structure in which call control is separated from service bearer. As shown in Figure 1-12, call control is separate from the media gateway (MG) and is processed by the media gateway controller (MGC). Therefore, MGCP is in nature a master-slave protocol. The MG establishes various service connections under the control of the MGC.
Figure 1-12 MGCP Master-Slave Control
H.248 Protocol
H.248 is the same type of protocol as MeGaCo and completed by the ITU-T and IETF together, used as a media gateway control protocol between an MGC and an MG. It takes the place of MGCP. H.248 features the following:
  • Functions on the basis of MGCP and therefore it inherits all advantages of MGCP.
  • Works in master-slave mode.
  • Uses binary coding or text coding for H.248 messages. MGC must support these two coding modes and MG supports either of them.
  • Uses User Datagram Protocol (UDP), Transmission Control Protocol (TCP) or Stream Control Transmission Protocol (SCTP) (IP-based signaling transmission) for underlying transmission.
Compared with MGCP, H.248 has the following advantages:
  • Supports voice services and multi-media connections.
  • Supports text coding and binary coding.
  • Features expandability.
SIP Protocol

Session initiation (SIP) is a session control protocol running at the application layer, which sets up, modifies, and terminates a session. A session can be an application, such as multi-media conference and Internet call.

Comparison Between the H.248 Protocol and SIP Protocol
Table 1-5 describes comparison between the H.248 protocol and SIP protocol.
Table 1-5 Comparison Between the H.248 protocol and SIP protocol

Item

SIP Protocol

H.248 Protocol

Standard

IETF/TISPAN

ITU_T/TISPAN

Architecture

Distributed/Intelligent clients

Centralized/Dumb end_point

Call control

Proxy/Redirect Server

Call agent/MGC

Transport protocol

UDP/TCP/SCTP

UDP/TCP/SCTP

Multi-media service supported

Yes

Yes

Supplementary service

Provided by endpoints or by call control

Provided by call control

ISDN service

Not defined in TISPAN R1

Use IUA Support

Separation of Media and Signaling Streams

Separation of media and signaling streams indicates that signaling streams (H.248/SIP) and media streams can be transmitted upstream to different virtual private networks (VPNs) through different IP addresses and VLANs. The separation facilitates network planning and meets control requirements, as shown in Figure 1-13.
Figure 1-13 Separation of media and signaling streams
Application scenarios of separation of media and signaling streams:
  • Signaling and media flows are transmitted upstream to different VPNs and they use different control policies, such as QoS.
  • Different user groups use different signaling IP addresses/VLANs and media IP addresses/VLANs. This meets the special monitoring requirements.

External Direction of Voice Media Streams and Signaling Streams

Voice services can be classified into the following 3 scenarios in which media streams and signaling streams are transmitted in different directions.
  • Communication under different softswitches/IMS
  • Communication under the same softswitch/IMS but different access gateways (AGs)
  • Communication under the same softswitch/IMS and same AG
The following shows directions of voice media streams and signaling streams in these 3 scenarios:
Communication Under Different Softswitches/IMS

Communication Under the Same Softswitch/IMS but Different AGs

Communication Under the Same Softswitch/IMS and Same AG

Internal Direction of Voice Media Streams and Signaling Streams

POTS Access Mode

The following flash video demonstrates the directions of media streams and signaling streams inside an AG device in POTS access mode.

Functions of each module are as follows:
  • The subscriber line interface circuit (SLIC) processes analog signals, including feeding phone sets, sending voice frequencies to phone sets, and generating ringing, as well as detecting off-hook, pulse dialing, on-hook, and hookflash signals.
  • The coder/decoder (CODEC) converts between analog and digital signals. It converts analog signals to digital signals in the upstream direction and digital signals to analog signals in the downstream direction.
  • The digital signal processor (DSP) supports the following functions:
    • Codes and decodes voice signals. The DSP encapsulates the digital signals sent by the CODEC into VoIP packets in the upstream direction and restores the VoIP media streams transmitted over the GE bus to digital signals in the downstream direction.
    • Manages the SLIC and CODEC using the SPI bus.
  • The GE LSW and voice processing module process signaling streams and media streams. They determine whether to discard or forward signaling/media streams based on the IP address of the signaling/media streams and the UDP port number. The voice processing module is a daughter board on the control board.
ISDN Access Mode

The following flash video demonstrates the directions of media streams and signaling streams inside an AG device in ISDN access mode.

Functions of each module are as follows:
  • Network termination (NT) 1: functions similarly as the physical layer of the OSI reference model. The functions of NT1 are associated with inherent physical and electrical characteristics of the network.
  • Terminal adapter (TA): Non-ISDN terminals do not support channel-D functions. To connect a non-ISDN terminal to the ISDN network, a TA is required.
  • ISDN chip group: connects to NT1 using a U interface for ISDN BRA services. The ISDN chip group converts analog signals to digital signals and implements codec of a U interface (such as 2B1Q and 4B3T codes).
  • E1 chip group: connects the E1 interface to NT 1 for the ISDN PRA service. It converts analog signals to digital signals.
  • DSP: codes and decodes voice signals. The DSP encapsulates the digital signals sent by the ISDN or E1 chip group into VoIP packets in the upstream direction and restores the VoIP media streams transmitted over the GE bus to digital signals in the downstream direction.
  • The GE LSW and voice processing module forward signaling streams and media streams. They determine whether to discard or forward signaling/media streams based on the IP address of the signaling/media streams and the UDP port number. The voice processing module is a daughter board on the control board.
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Updated: 2019-02-22

Document ID: EDOC1100067358

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