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1 Voice Over IP, and Voice Applications Eyal Wirsansky, Verso Technologies JaxJUG
2 Analog Telephony Mr. W AG Bell X
3 What the is aa Switch??
4 Moving to Digital Voice (TDM) Separation of Voice and Signaling Signaling using data messages Voice is digitized into a bit stream Voice channels are multiplexed
5 Digitizing Voice (PCM, G711) amplitude time Filtering g( (3kHz) Sampling (8Khz, 1/125μs) Quantizing... 1 byte 1 byte 1 byte (8 bits, 256 values) => 64 Kbit/sec
6 A-Law and Mu-Law
7 Time Division Multiplication (TDM) Frame 24 time slots = T1 32 timeslots = E1
8 Circuit Switched / TDM Network Telephone Signaling Telephone CO Switch Circuit is nailed up for duration of the call Entire bandwidth of time slot is always available during call No real possibility of congestion once time slot is acquired 1 Frame CO Switch All available bandwidth is consumed All available bandwidth is consumed whether there is traffic or not
9 IP Telephony Voice and signaling g are separated, but both are carried over data / IP network Voice traffic is carried as individually addressed packets, that may or may not follow the same path, depending on network conditions Bandwidth is consumed only if/when data is sent Bandwidth is shared, so congestion must be managed
10 TDM Bandwidth Utilization
11 Packet-Switched Bandwidth Utilization i
12 Why VoIP? Network convergence Better bandwidth utilization Use of Codecs Cheaper or free transport General Purpose platforms Diversity, Flexibility
13 VoIP Components Endpoints: SIP phones (soft/hard) IADs Gateways: Signaling Gateway (Softswitch) Media Gateway Access Gateway Service Nodes: Application Servers Media Servers AAA platforms
14 VoIP Protocols SIP / SDP - Signaling RTP / RTCP Media (Voice) MGCP et al. Media Control RADIUS, DIAMETER (AAA)
15 VoIP Components and Protocols * 8 # Access Gateway
16 VoIP Components and Protocols * 8 #
17 VoIP Components and Protocols * 8 # * 8 #
18 VoIP Components and Protocols * 8 # * 8 #
19 The SIP Protocol ACK SIP/2.0 Via:SIP/2.0/UDP station1.work.com To: Cseq:1 ACK Content-Length:0 INVITE SIP/2.0 Via:SIP/2.0/UDP station1.work.com To: Cseq:1 INVITE Content-Length:167 Content-Type:application/sdp (message body) SIP/ Ringing Via:SIP/2.0/UDP station1.work.com To: Cseq:1 INVITE Content-Length:0 (message body) SIP/ OK Via:SIP/2.0/UDP station1.work.com To: ork Cseq:1 INVITE Content-Length:109 Content-Type:application/sdp (message body)
20 Challenges of VoIP Availability ( five nines ) Quality of service packet loss, packet timing Security Firewalls, NAT
21 Use Cases of VoIP Toll Bypass (Calling Cards) Residential VoIP Soft Phones services (FWD, Skype) Enterprise Convergence Voice Applications (IVR driven)
22 Examples of Voice Applications Auto-Attendant, Voic Prepaid Services Calling Card Conferencing Self Service Applications Automated Banking, Healthcare Weather Service, Stock Quotes Voice Portal
23 VoIP/Voice Applications i Data Source * 8 #
24 VoiceXML Based Solution * 8 #
25 VoiceXML W3C standard XML specifying dialogues between a human and a computer Voice applications are analogous to HTML Oft i l S h S th i d Often involves Speech Synthesis and Speech Recognition
26 VoiceXML Examples Tell Me Voice Portal TELL (8355) TELLME-1 Many others: VOXEO Nuance/Speechworks BeVocal Cafe
27 Using Wireshark Wireshark is a free network protocol analyzer ( ) Runs on most computing platforms Captures all packets seen by the network interface card Analyzes, filters and displays packets, with numerous protocols supported
28 The Asterisk Platform Open source/free software implementation of a telephone private branch exchange (PBX) Originally created by Mark Spencer of Digium. Written in C++, runs on Linux platforms Combines Signaling capability (both TDM and IP) with media capabilities. Contains Pre-packaged applications, such as Voic and Conferencing.
29 Asterisk Resources Home Page: Official Book (Free PDF Download): Asterisk in a Box : VoIP Wiki
30 Voice Applications for Asterisk Option 1: definitions in extensions.conf file info org/wiki Option 2: Asterisk Gateway Interface (AGI) Utilizes stdin, stdout for interfacing Launches and communicates with an external script (Perl, Python, bash, etc.) Option 3: FAST AGI and/or Manager API Utilize TCP interface Allow commands, responses and events
31 Asterisk-Java a Set of Java classes Allow building Java applications that interact with Asterisk via the Fast AGI and/or the Manager API Can run on external platform Java Application Server asterisk-java.org
32 * 8 # Asterisk-Java Demo Prompt Storage NFS Harmony 6000 Media Server AAA Server AAA Interface MGCP Asterisk-Java Application Server SES Framework One-Stag Application pp Future Applications Calling Card Application SES Oracle Fast-AGI SIP RTP Asterisk Soft Switch Signaling Media Gateway Media SES Setup and Management Apache PSTN Network SIP RTP End User
33 Wireshark Captures for the At Asterisk-Java ikj Demo The next slides show packets captured by Wireshark during the Asterisk-Java demo; each slide uses a different filter: SIP shows the signaling between the X-Lite SIP phone and the Asterisk platform TCP port 4573 shows the Fast-AGI protocol communications between the Asterisk-Java Application Server and the Asterisk Platform
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