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4-22696782-021 سمینار تخصصی. 4-22696782-021 What is PSTN ? (public switched telephone network) تیرماه 1395.

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Presentation on theme: "4-22696782-021 سمینار تخصصی. 4-22696782-021 What is PSTN ? (public switched telephone network) تیرماه 1395."— Presentation transcript:

1 4-22696782-021 سمینار تخصصی

2 4-22696782-021 What is PSTN ? (public switched telephone network) تیرماه 1395

3 4-22696782-021 The Beginning of the PSTN

4 4-22696782-021 Centralized Operator: The Human Switch

5 4-22696782-021 Analog and Digital Signaling

6 4-22696782-021 Analog and Digital Signaling

7 4-22696782-021 Meshed Network Versus Hierarchical Network

8 4-22696782-021 Circuit-Switching Hierarchy

9 4-22696782-021 PSTN Signaling User-to-network signaling This is how an end user communicates with the PSTN. Network-to-network signaling This is generally how the switches in the PSTN intercommunicate.

10 4-22696782-021 ITU-T International Numbering Plan  ITU-T Recommendation E.164 specifies that a Country Code (CC), National Destination Code (NDC), and Subscriber Number (SN) be used to route a call to a specific subscriber.  The CC consists of one, two, or three digits.  NDC and SN vary in length based on the needs of the country. Neither one has more than 15 digits.

11 4-22696782-021 The architecture built for voice is not flexible enough to carry data. Drawbacks to the PSTN Data/Voice/Video (D/V/V) cannot converge on the PSTN as currently built. The PSTN cannot create and deploy features quickly enough. Data has overtaken voice as the primary traffic on many networks built for voice.

12 4-22696782-021 Circuit Switching Versus Packet Switching

13 4-22696782-021 SS7 Network Architecture

14 4-22696782-021 Associated Signaling

15 4-22696782-021

16 Quasi-Associated Signaling

17 4-22696782-021 SS7 Protocol Stack Versus the OSI Model

18 4-22696782-021 Network for Call Setup and Teardown

19 4-22696782-021 OSI Reference Model

20 4-22696782-021 VoIP delay or latency is characterized as the amount of time it takes for speech to exit the speaker's mouth and reach the listener's ear. Three types of delay are inherent in today's telephony networks: propagation delay, serialization delay, and handling delay.

21 4-22696782-021 End-to-End Delay

22 4-22696782-021 Jitter jitter is the variation of packet inter arrival time.

23 4-22696782-021 ITU-T Codec MOS Scoring

24 4-22696782-021 Echo echo is normally caused by a mismatch in impedance from the four-wire network switch conversion to the two-wire local loop

25 4-22696782-021 H.323  H.323 is an International Telecommunication Union Telecommunication Standardization Sector (ITU-T) specification for transmitting audio, video, and data across an Internet Protocol (IP) network, including the Internet.  The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multipoint conferences.  Currently, H.323v5 (version 5) is considered the latest version as ratified by ITU.

26 4-22696782-021 H.323 Elements

27 4-22696782-021 Relationships of H.323 Components

28 4-22696782-021 Elements of an H.323 Gateway

29 4-22696782-021 Layers of the H.323 Protocol Suite

30 4-22696782-021 Direct Endpoint Signaling Same Gatekeeper

31 4-22696782-021 SIP

32 4-22696782-021 Functionality That SIP Provides User location SIP provides the capability to discover the location of the end user for the purpose of establishing a session or delivering a SIP request. User mobility is inherently supported in SIP User capabilities SIP enables the determination of the media capabilities of the devices that are involved in the session. User availability SIP enables the determination of the willingness of the end user to engage in communication. Session setup SIP enables the establishment of session parameters for the parties who are involved in the session. Session handling SIP enables the modification, transfer, and termination of an active session.

33 4-22696782-021  User agent A user agent (UA) is a logical function in the SIP network that initiates or responds to SIP transactions. A UA can act as either the client or the server in a SIP transaction. A UA might or might not directly interact with a human user. A UA is stateful that is, it maintains session or dialog state.  User agent client A user agent client (UAC) is a logical function that initiates SIP requests and accepts SIP responses. Examples of UAC are a SIP phone initiating a call on behalf of a human user or a SIP Proxy forwarding a request on behalf of a UAC.  User agent server A user agent server (UAS) is a logical function that accepts SIP requests and sends back SIP responses. A SIP phone accepting an INVITE request is one example.  Proxy A proxy is an intermediate entity in the SIP network that is responsible for forwarding SIP requests to the target UAS or another proxy on behalf of the UAC. A proxy primarily provides the routing function in the SIP network. A proxy might also enforce policy in the network, such as authenticating a user before providing him with service. A proxy can be stateless, transaction statefull, or call statefull. Typically, proxies are transaction stateful that is, they maintain state for the duration of a transaction (about 32 seconds).  Redirect server A redirect server is a UAS that generates 300 class SIP responses to requests it receives, directing the UAC to contact an alternate set of Uniform Resource Identifiers (URI).  Registrar server A registrar is a UAS that accepts SIP REGISTER requests and updates the information from the request message into a location database.  Back-to-back user agent A back-to-back user agent (B2BUA) is an intermediate entity that processes incoming SIP requests as a UAS. To answer the incoming SIP request, the B2BUA acts as a UAC, regenerates a SIP request, and sends it on the network. A B2BUA must maintain dialog state and participates in all transactions within the dialog. SIP Network Elements

34 4-22696782-021 Proxy Mode of Operation

35 4-22696782-021 Redirect Server Mode of Operation

36 4-22696782-021 B2BUA Server Mode of Operation

37 4-22696782-021 SIP Call Setup and Teardown Involving the SIP Proxy

38 4-22696782-021 THANK YOU داده پردازان ایتوک نماینده رسمی NEC در ایران


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