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ITCV 1033 Intro to Voice & Data Networking

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Presentation on theme: "ITCV 1033 Intro to Voice & Data Networking"— Presentation transcript:

1 ITCV 1033 Intro to Voice & Data Networking
VoIP ITCV 1033 Intro to Voice & Data Networking Glenn Jones 2015

2 Introduction to VoIP Voice conversations have typically been carried across the PSTN PSTN is a WAN (wide area network) WAN’s use different technolgies than LAN’s By digitizing and packetizing voice, it can be carried across a data network, such as Internet A voice application over IP is called VoIP Glenn Jones 2015

3 Review Question Explain the meaning of VoIP Glenn Jones 2015

4 Voice Over the Internet
At first, data was modulated to look like voice Now, voice is being packetized to look like data 75-90 % of a voice conversation is idle time Wasted bandwidth and revenue for telco PSTN boasts % uptime! This will be difficult to approach with VoIP Glenn Jones 2015

5 Review Question What was one of the major accomplishments of the PSTN in its 100+ year history? Glenn Jones 2015

6 Voice Over Internet Protocol
The PSTN boasts % uptime It will be difficult to achieve this reliability with VoIP Fax appears to the PSTN as an analog device Data transmission via facsimile FoIP is very popular in many industries, such as health care Many companies use VoIP for internal calls QoS can be achieved across internal data networks POTS will continue to be used for many years Glenn Jones 2015

7 Review Question Define and describe FoIP Glenn Jones 2015

8 Common PSTN Features CLID—calling line identification
3-way and conference calling Call transfer Voice mail Voice to text VoIP must at least match these features Glenn Jones 2015

9 Quality of Service Voice traffic cannot tolerate significant delays
Voice traffic must be given priority over data Methods of achieving acceptable QoS Int-Serv using RSVP (resource reservation protocol); each flow is maintained; not scalable Diff-Serv using DS bits on each datagram; scalable IEEE 802.1p standard for layer 2 LAN’s Glenn Jones 2015

10 Review Question Define QoS and explain its significance in VoIP
Glenn Jones 2015

11 Real-Time Transfer Protocol
RTP carries the data (20ms of packetized voice) RTCP (control) monitors flow QoS on different port 20ms of voice is the RTP payload (application) RTP hdr—time stamp, sequence, compression info UDP header contains source & destination ports IP header contains source and dest IP addresses Ethernet header contains source & dest MAC addr Glenn Jones 2015

12 Review Question Approximately how much voice conversation is carried in each RTTP packet? Glenn Jones 2015

13 Delay Max 50ms (1/20th second) acceptable in PSTN
VoIP packets may be buffered until all arrive Jitter = variation in datagram arrival times Because of buffering, more delay is acceptable with VoIP than with POTS; up to 150 ms Glenn Jones 2015

14 Review Questions Compare acceptable delay ~ PSTN & VoIP
Describe jitter Glenn Jones 2015

15 Voice Gateways Interfaces analog POTS devices with IP networks
PSTN IP Network Interfaces analog POTS devices with IP networks Works for analog telephones and analog faxes AKA voip gateway, broadband gateway, IP gateway AKA media gateway (may include video media) AKA analog telephone adapter (ATA) VoIP phones have internal gateway functionality Glenn Jones 2015

16 Review Question What is a voice gateway? Glenn Jones 2015

17 VoIP Signaling Just as in-band and SS7 signaling are used with PSTN, VoIP calls must be set up & managed by MGCP—Media Gateway Control Protocol MEGACO—new name for MGCP Emphasis on standardizing industry equipment Developed by ITU and IETF Includes H.248 protocol H.323 SIP Glenn Jones 2015

18 VoIP Signaling Protocols
Service providers are “in control” with: MGCP & Megaco Protocols developed by / for telco Master/slave; call agent or softswitch is required End users are “in control” with these protocols H.323—dev by ITU in 1990’s; multimedia, robust SIP—developed by IETF in late 1990’s for voice SIP & H.323 are peer-to-peer protocols Glenn Jones 2015

19 Review Question Compare Megaco with H.323 and SIP protocols
Glenn Jones 2015

20 H.323 Application Protocol
Circuit switched PSTN calls have 64kbps VoIP datagrams are compressed to conserve BW Silence is removed, results in <10kbps bandwidth “H” series standards are from the ITU-T H.323 developed for multimedia—voice & video Other H protocols deal w/ signaling, security, etc. Based on session oriented TCP (overhead & delay) Uses RSVP based on flows (conversations) Glenn Jones 2015

21 H.323 Networks Required components: Optional components:
H.323 terminal—computer, phone, etc. H.323 gateway—protocol converter (POTS IP) Gateway function may be included in intelligent terminals Optional components: H.323 gatekeeper—authentication, management Aka softswitch, voice server H.323 multipoint control unit; conferencing >2 users G W G W Term Term IP Network Glenn Jones 2015

22 H.323 Networks With optional components:
Gatekeeper—deals with management functions offloaded from gateways in larger H.323 networks MCU for audio / video conferencing Gatekeeper / softswitch / voice server MCU G W G W Term IP Network Term Intelligent Terminal Intelligent Terminal Glenn Jones 2015

23 SIP Application Protocol
Session Initiation Protocol Developed by IETF—I’net engineering task force Developed as a multimedia alternative to H.323 Less complicated than H.323 SIP is a peer-to-peer (client / server) protocol End devices are UAC (user agent client) or UAS (server) Depends on which device initiated the call Can use TCP or UDP at layer 4 Glenn Jones 2015

24 SIP Devices UA (user agent)—IP phone, softphone
UAC (client) initiates (requests) the call UAS (server) accepts (responds to) the call Proxy server—makes requests for others Aka softswitch, voice server Redirect server—maps and returns addresses Registrar server—location database Glenn Jones 2015

25 SIP Devices UA (user agent)—IP phone, softphone
UAC (client) initiates (requests) the call UAS (server) accepts (responds to) the call Proxy server—makes requests for others Aka softswitch, voice server Redirect server—maps and returns addresses Registrar server—location database SIP redirect server SIP registrar server UA IP Network Proxy server / softswitch / voice server UA Glenn Jones 2015

26 Review Question What are 2 other names for the gatekeeper (H.323) / Proxy server (SIP) Glenn Jones 2015

27 RTSP Real Time Streaming Protocol
Allows listening to downloads on the fly No need to wait until entire file is transferred Glenn Jones 2015

28 Call Quality and Integrity
Lost data packet can be retransmitted Lost voice packet will leave space in audio VoIP devices will need security updates Authentication may be accomplished by Encryption—resource intensive; slows process IPSec Challenge-response based authentication Glenn Jones 2015

29 VoIP Security Firewalls filter packets based on addresses, ports, content, etc. Firewalls close unused ports to prevent attacks Corporate firewalls must open the correct ports VoIP traffic can use multiple, varying ports Allowing VoIP traffic can be very complex Glenn Jones 2015

30 BW & Compression Standards
G.711—uncompressed PCM at 64kbps Codec—coder / decoder Converts analog to compressed digital signal G.723.1—compressed to 5.3kbps G.729—compressed to 8kbps Overhead is required Bandwidth depends on compression & overhead Glenn Jones 2015

31 Review Questions What is the data rate for uncompressed digital voice?
Why is it desirable to compress voice packets? Glenn Jones 2015

32 Traffic Engineering Analogy—Municipal highway system VoIP
Traffic flows freely most of the time Busy hours 8AM and 5PM More lanes are better, but at what cost? VoIP Plenty of network bandwidth most of the time Congestion during busy hours Design system for optimum performance Glenn Jones 2015

33 Mean Opinion Score & Delay
5 Perfect 4 Fair (PSTN quality) G.711’s MOS=4.3 3 Annoying G.723.1’s MOS=3.9 2 Horrible 1 Impossible to communicate MOS developed by ITU to grade sound quality Up to 150 ms delay is acceptable with VoIP Sources of delay include Processing , algorithmic, packetization, buffering, switching, etc. De-jitter delay; wait until all packets are in buffer Glenn Jones 2015

34 MOS—Mean Opinion Scores
5 - Perfect. Like face-to-face conversation or radio reception. 4 - Fair. Imperfections can be perceived, but sound still clear. This is (supposedly) the range for cell phones. 3 - Annoying. 2 - Very annoying. Nearly impossible to communicate. 1 - Impossible to communicate Glenn Jones 2015

35 Review Questions Define MOS Give an example of a conversation w MOS 5?
What is the MOS of most cell phones? What is the MOS of analog PSTN conversations? Glenn Jones 2015

36 Business Considerations
ROI (return on investment) depends on Hard benefits—cost savings Soft benefits—increased productivity Expenses Hardware and software IP phones or softphones Network upgrades Labor—installation, training, maintenance, contracts Glenn Jones 2015


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