Netizen's VoIP summary 1
---- Gateway: Code and decoding voice, implement protocol conversion on the PSTN side and IP side; ----- Netkeeper: Provide call control services for gateways and terminals to implement regional management , Access control, address parsing, bandwidth management, etc .; ----- AAA server (Authentication, Authorization, Account): Responsible for authentication, authorization, and billing for users; ----- Telephone terminal: IP phone terminal Traditional voice phone, PC, IP phone, can also be a multimedia service terminal integrating speech, data, and images. ---- H.323 Definition Description How to implement multimedia communication systems on grouping networks without QoS assurance; applicable services are multimedia communications including voice, data, and video and their combinations. ----- H.323 system defines the following entities: Terminal; Gateway; GK); MCU (including MC and MP); terminal, gateway and MCU are collectively referred to as endpoints, Run and accept calls; GK is responsible for managing other entities to which they registered, together into a region (zone); the area is independent of the network topology, which can be constructed from a plurality of network segments connected by routing devices. Multiple regions constitute a H.323 system ------ Main factors affecting QoS (1) Call establishment delay: Network delay user authentication link established database query and processing PSTN side processing voice transmission : Network Transmission Temperature Network Confusion, Decoding Decoding Decoding Decodation ------- The main factor affecting QoS (III Packet loss rate caused by two Jitter Buffer: Network transmission packet loss network Congestion Time Gateway Equipment Active Packet Voice Quality: The Code Algorithm Adopted Simultaneous Loss L Signals on the Network Date of Data Traffic Sampling and Error Correction Techniques: Allow Telephone and Network and Other Phone Systems Communication The unit notification, dial tone, ringing, and busy signals are signaling used in analog phones. L interface type and signaling method 1: Router's voice interface type: ------ FXS: RJ-11 , 2-wire interface ------ fxo: RJ-11 module socket ------ E & M: 2- line, 4-line, Cisco supports E & M type I, II, III, V in its VoIP products 2: FXO / FXS signaling technology ------ loop start ----- Ground start l Factors affecting voice quality ---- Compression ---- Voice activity detection VAD: Detection session Mute The time period and the generation of data streams at those time periods. ---- Echo: is caused by electrical reflection in the voice network, which is usually between 4-wire switches and 2-wire switch local loops. Impedance difference. Method for processing echo: First, reduce the power of the signal, so that the volume of the echo is the smallest. The second is to use echo cancelor. ----- Time: propagation delay and processing delay ------ - Jitter: Various delay changes lead to changes in data packets in the network.
Set a buffer to compensate for jitter ---- Lost Packet: CODEC ----- Switch Switch: Rebate, Delay IP Data Stream Service Q (Compressed) RTP: Configuration IP of Conference Layer Protocol CRTP rtp header-compression passive ip rtp compression connections n frame-relay ip rtp header-compression [passive] frame-relay map ip add dlci [broadcast] rtp header-compression active // multipoint interface or sub-interface queuing l // l PPP Enhancement: Segmentation and Cross LFI: Objective To improve the efficiency of transmitting large packets on multiple parallel links, in order to transfer time-sensitive data, you can use the IP RTP RESERVE command to configure a special queue, which only contains RTP data flow configuration: nultilink virtual-template 1 interface virtual-template 1 ip add 192.168.1.1 255.255.255.0 ppp multilink ppp multilink interleave ppp multilink fragment-delay 20 ip rtp reserve 16384 100 384 interface s0 / 0 bandwidth 512 encap ppp no ip Add PPP MultiLink MultiLink-Group 1 L Data Streaming L RSVP: The RSVP sender starts this process by sending an RSVP band with the reception. l IP priority L Weighted random detection WRED Weighted RED For different IP priority-based service classes and RSVPs provide different packet discarding thresholds. INT S0 / 0 IP Add Random-Detect voice port and dial peer L voice port L Physical interface and signaling L Analog speech port: 3 types of signaling can be used to simulate system 1. FXS 2. FXO 3. E & M: I IIII V 2- Line, 4-Line L FXS and FXO interface FXS: Line voltage, ringing signal generation, off-hook detection, call process indication (ring, busy signal) and identify dial numbers for routing. L E & M Interface Monter Signaling: Signal [Wink-Start | Immediate | Delay-Dial] Dialing Method: DTMF or PULSE: Dial-Type [DTMF | PULSE] Call process tone is normalized in the geographic area: CPTONE Region The E & M interface can be 2-wire or 4-line: Operation {2-Wire | 4-Wire} E & M interface type: Type {1, 2, 3, 4, 5} The terminal impedance of the voice port must be properly configured, Prevent unwanted echoes, mismatch impedance settings will cause the speech signal to reflect at the interface, generate a large number of echo: IMPEDANCE: private line automatic ringing Plar line can also be established on E & M interface, once detected a peak state configuration Automatically connect to a fixed destination for the voice port of Plar.
: Connection Plar String L Other voice port parameters: 1. Gain and loss parameters: Adjust the voice port command for the input gain is: Input Gain N, the command of the control signal is reduced in the output speech port is: Output Attention N 2. Echo cancel: Echo is basically a VoIP summary from a netizen 2
