VoIP small knot
Factors affecting VoIP voice packet network:
1. delay
a) transmission delay
Determined by the transmission medium
b) Processing delay
Delay generated by the network device forwarding frame
In the Cisco IOS IP phone, the two millisecond delays are generated every two, and the DSP group will be generated in the same division of G.729. So a packet produces a 25 milliseconds to generate a voice sampling sample every 10 milliseconds. Delay.
Cisco constructs frames by using DSP, without processing of frames by routers, can reduce the load of the router.
c) Queue delay
The queue strategy in the network is determined.
ITU-T G.114 recommends that the end-to-end delay is 60 delays as high as 2 seconds. The queue delays the quality of the words, the time delay of the one-way end peer is not more than milliseconds. Therefore, the time delay of the network 90 millisecond is just part of the end-to-end delay, and the other is 150 milliseconds. Cisco's voice device can be back. In not managed, congested networks, queuing time is trembling.
2. Tremble
Definition: The grouping to the destination interval change. Serious trembows will cause the total delay of the network.
Measures: Use a trembling buffer, which, according to the time difference of the RTP packet arrival time, the number of times that enters in the buffer, can dynamically reduce or increase the concept of a software. This queue can grow or shortened accordingly when it is. According to the package, the size of the duration buffer is to compensate for the magnitude of the buffer caused by the trembling, or a dynamic queue enters the data of the buffer according to the adjacent datagram. The speed is the same, when the data is delayed in the network.
3. Voice compression
PCM (Pulse Code Modulation): Coding the amplitude of 8 to 12 bitmine dialogue.
ADPCM (Adaptive Difference Pulse Code Modulation): The amplitude difference between 4 ratios is used, and the rate of change of the amplitude is encoded.
G.711 ---- 64kbps PCM voice coding technology
G.726 ---- ADPCM encoding of 40, 32, 24, and 16kbps.
G.728 ---- Description CELP (encoded excitation linear prediction compression) speech compression of a low delay variant.
G.729 ---- Description You can compress the voice into a CELP compression criterion that compresses the 8kbps stream. Where G.729 and G.729A
You can provide a voice quality with 32 kbps adpcm.
G.723.1 ---- Describe a compression technology, as a speech or other audio signal of the H.324 standard multimedia service. An integral part of the quasi-series, we can use very low bit rates
4. echo
Definition: By impedance mismatch or other reasons, causing the voice from the user to be bounced.
Solution: Echo canceller, reverse mirror, and to eliminate possible echo users to make a speech reverse mirror save a period. Along, listen to the sound from the peer user and subtract
Key parameters: "Echo tail" causes echo; if it is too large, the elimination refers to the recovery canceller to give a longer time to aggregate and eliminate echoes. total time. If the configuration is too small, it will cause easy
5. Packet loss
Definition: If the system does not receive the last packet to be broadcast once again. . The expected voice group (expectation time is variable due to the packet loss is just 20 milliseconds, one), can be considered to be lost, and the same listener will not pay attention to the different voice quality
Measures: "Compensation Strategy", when a group is lost and trembling is that the listener cannot hear the silent gap. When the buffer is full, the acceptor will replay the last received group,
6. Voice activity detection (VAD) 7. Digital-to-analog conversion
8. Serial encoding
Definition: In the voice transfer, after multiple decompression, G.729 can tolerate two compression / decompression processing, and there are more cases in the case of G.723.1. Resulting in a decline in speech quality. Different encoders have different reactions to serial encoding. The robustness of multiple compression processing is not high. This phenomenon dials in centralized management
Solution: Use GateKeeper or Call Manager
9. Transfer Protocol
RTP / RTCP / RUDP
What is RTP?
First, the IP network is not a variety of applications that are sensitive to time, requiring one frequency or voice data. However, it does not guarantee the process of the appropriate transmission; in the packet protocol, it is running on UDP, which allows a network to provide multiple concurrent concurrent architecture, and will have retransmission, loss mechanism. Maintain synchronization. Transmission of Real-Time TRANSP, it mainly provides two key-free post-stamps, making the data recipients when connecting, not using the reserved UDP port, without conflicting. Package, charter, etc. For voice and video, etc., to transmit the visibility on IP network: Place the serial number on each packet to prevent the size of the trembling buffer. RTP is not a transfer layer, but each connection uses its respective port numbers.
What is RTCP?
RTCP is the control section and gateway support in the RTP data, such as audio and video network QoS feedback, and the real-time bridge of different media, RTCP's real-time bridges on the Internet, and more broadcast to single-point transmission conversion. Conference is supported. This support includes a source identification. RTCP also provides recipients to multipartboard
What is RUDP?
Reliable user transport protocol, it implements the reliability of transmission, redundant redundancy, which can simultaneously send multiple identical packets, but do not have to be based on the connection-oriented protocol, such as TCP. It can achieve a loss.
10. Dial plan design
supplement:
It can be easily understood in the application:
FXO is an ordinary telephone interface that requires remote feed;
The FXS interface is the inner extension interface of PBX, feeds remotely;
E & M is generally used in the PBX trunk interface.
If the E & M interface board is configured on the PBX, it is the same as the E & M interface, which is the same as Cisco;
The FXS interface is used to connect the phone, the signal; the normal trunk connection of the PBX can also be supplied to the PBX trunk with the FXS interface of the Cisco device.
The FXO interface is used to connect PSTN, and the user PBX analysis, which is more connected to the PBX inner extension for the bureau, and is also flexible by PBX. The FXO interface provides a feed signal. The above is for use
H.323 protocol stack
One. Software composition
Function protocol
Call signaling H.225
Media Control H.245
Audio encoder G.711, G.722, G.723, G.728, G.729
Video encoder H.261, H.263
Data sharing T.120
Media transmission RTCP / RTP
two. System consisting of system
1. terminal:
Provide audio and multi-point meeting audio, as well as optional video-based video interfaces, and packet-based network interfaces. Frequency and data. The H.323 terminal must have a system control unit, a media transmission,
2. Gateway:
Responsible for converting audio, video and data, and SCN call setup and revoking. The transmission format between communication systems and the conversion between the communication system and the communication protocol, including IP network 3. Net warning
Provide each call and a call level control service to the H.323 endpoint. It needs to complete the following features:
Address Conversion - Provides H.323 alias and E.164 addresses to IP address services.
Authorization Control - Use the ARQ / ACF / ARJ message to provide authentication access to H.323.
Bandwidth Control - Managing endpoint bandwidth requests by using BRQ / BCF / BRJ messages.
Time Zone Management - Terminal, Gateway, and MCU for registration.
4. MCU
The MCU is an end point that supports multi-point meetings and talks between three or more endpoints in the conference. It can control the included MC and one or more MPs. MC: Multi-point controller, support multi-point conference capability of multi-point endpoints.
MP: Multi-point processor, accept video / audio / data stream, and distributes these transport data to the end point to participate in multi-point meetings.
5. H.323 proxy server
It mainly plays a compensating role, change, route separation, and firewall functionality provides security for some of the functional incomplete H.323 terminal H.323 data. Provide a complete function. For example: RSVP, address transfer
three. H.323 protocol
1. Registration, Licensing, and Status Signaling (RAS):
On the H.323 based on Gatekeeper, it opens the execution registration, license, bandwidth change, and the control of the call turn-on before the establishment of any other channel is created. Yes, it is independent of the call control signaling and the media transfer state, and the RAS message of the release process. Channels established between the endpoint and the gatekeeper on the IP network. Rely on UDP non-reliable transmission protocol to transmit
Neighborhood Discovery: Divided into two ways, manual and automatic; manual shifts to send multicast (224.0.1.41) through the end point (224.0.1.41) to discover the gatekeeper. By setting the IP address of the gatekeeper, the purpose of registering on the gatekeeper; from the UDP discovery port of the gatekeeper is 1718, the registration and status port number is 1719
2. Call Control Signaling:
The connection between the endpoints, dimensionally creates a connection, maintenance, and revoking call across IP networks on the TCP1720 port. But port. Protective and broken chains. Based on H.225 Signaling, this proposal control channel. This port is the actual call control and maintaining the activity message in two ends in the actual call control and supporting the use and support of the Q.931 signaling protocol. The endpoint initiates the q.931 call control information, and then transferred to temporary after the initial call is established.
Signaling process
3. Media control and dissemination:
Provides a reliable H.245 channel for transport media control messages. General information. Use dynamic port creation links on IP and have multiple UDP streams to transmit. H.245 Processes the end to the end of the H.323 entity and the end point of the end point, how many H.245 channels will be created.
Control content:
Ability Exchange: Speech parameters for each end point, such as G.729, sampling rate, etc.
The master-slave termination: conflicts between the endpoints, especially when the two end points are the same operation, can determine that priority.
Round-trip delay: used to determine the delay between the source and destination endpoints.
Logical channel signaling: Opening and closing to determine how routing. Transfer channels of audio, video and data messages; establish two-way unidirectional channels; gatekeepers can be based on signaling
Quick connection process: You can make the base parameters, which is included in the H.225 signaling protocol, which can pass two simple messages, the media connection of this point-to-point call, if the H.225 message is included The start of the media is transmitted. By completion through a round-trip message. This is a parameter, then the hole is worn by H.245 after the H.225 negotiation: refers to the H.245 message can be encapsulated in the H.225 call signaling channel, rather than establish a separate H.245 control channel.
system structure
System call
SIP protocol
One. characteristic
1. Determine the location of the destination end point, support address resolution, name mapping, and callback redirect.
2. By using the SDP protocol to confirm the capabilities of each end point to ensure that each endpoint can be supported when multiple sessions are performed.
3. Determine the availability of the endpoint, and the reason that is not available.
4. Establish a link, if a underlying call is complete, the SIP creates an end-to-end connection between the two end points.
5. Handling the transfer and abortment of the call.
two. System Components
1. User Agent: SIP is an end-to-end protocol, which acts as a UAC, UAS role.
Ø UAC (User AgentClient): User Agent Client, issues a client of the SIP request.
Ø UAS (user agentserver): User agent server, accepts SIP request, and responds.
Ø In the SIP, there is no unpleasant as at the same time. The pair of clients and server, the same end point can be used as the server and client, but not
2. SIP gateway:
Ø Provide call control;
Ø Provide signaling conversion of SIP terminals and other types of terminals;
Ø Provide video and audio encoding;
Ø Call initiation and clearing (LAN and switching circuit network)
3. SIP server
Proxy Server: Accepting the Identity of the Client as the initiator of the request, and features. It is equivalent to the SIP request of the gatekeeper in H.323, forwarding the message forwarding to other ways to respond to the agent function as the same path. Before the server, the logo of the header is changed, and the generation is override, not the customer. At the same time, certification is available, authorized
Redirect Servers: Accept SIP requests and will include the next no processing or forwarding SIP requests. The redirect response of the hop server address is returned to the user. Do not accept calls, also
three. Architecture
1. SIP server: In addition to the agent and redirect server, there is a registration and positioning server.
Registering Server: On the SI instructor for moving, you can let other users can automatically register to the registration server inquiry locating server to find the terminal by checking the P user. For the positioning service, give the terminal's orientation information to register, the query of the positioning device is done by the proxy service.
2. Named: In the URL form to enter the address, SIP can name the maximum line terminal, such as the application of this Web mode. Form, through the DNS server to convert into IP
