Presentation on theme: "VoIP Voice Over IP Group 1: Mero Avanessian Tenghan Jiang Wendy Tran."— Presentation transcript:
VoIP Voice Over IP Group 1: Mero Avanessian Tenghan Jiang Wendy Tran
VoIP? ➢ Also known as: IP telephony (IPT), Internet telephony, broadband telephony, broadband phone service, broadband phone, internet phone service, digital voice ➢ What is VoIP? ➢ Short History ○ what we used to use ○ when it started ○ when it became popular
➢ packet switched protocol ➢ packet basics: ○voice signals (data), caller’s & receiver’s network addresses ○signal -> codec -> ADC -> RTP & encoding -> send packet ■packets can be sent from multiple routes ○RX -> RTP rearrange packets -> DAC -> sound card/phone How does it work?
➢ Minimum Bandwidth ○header usually takes 16kbps ○bandwidth depends on ■total packet size, codec bit rate, and number of packets sent How does it work?
➢ H.248 (megaco, gateway control protocol) ➢ H.323 ○ H.248 & H.323 recommended by International Telecommunication Union Telecommunication Standardization Sector (ITU-T) ➢ Real-Time Transport Protocol (RTP) ➢ Real-Time Transport Control Protocol (RTCP) ➢ Secure Real-Time Transport Protocol (SRTP) Protocols
➢ Media Gateway Control Protocol (MGCP) ➢ Session Traversal Utilities for NAT (STUN) ➢ Transport Layer Security (TLS) ➢ Session Initiation Protocol (SIP) ➢ Session Description Protocol (SDP) Protocols
➢ Call Quality ○ bandwidth ○ hardware equipment ➢ Consumer Market ○Low costs ➢ Corporate Use ○bandwidth efficiency and low costs ○run over a single network Adaptations
Companies that provide VOIP phones Cisco RingCentral Vonage Applications for Computers and Smart Phones Google Hangouts Protocols: STUN, TLS Skype Protocols: P2P ZoiPer Protocols: SIP, RTP Applications
Using Zoiper, a free VoIP softphone dialer. Linphone.org, a free SIP provider. 2 Cell phones running Android OS. CSULA wifi. WireShark Packet Decoding DEMO
INVITE = Establishes a session. ACK = Confirms an INVITE request. BYE = Ends a session. CANCEL = Cancels establishing of a session. REGISTER = Communicates user location (host name, IP). OPTIONS = Communicates information about the capabilities of the calling and receiving SIP phones. PRACK = Provisional Acknowledgement. SUBSCRIBE = Subscribes for Notification from the notifier. NOTIFY = Notifies the subscriber of a new event. PUBLISH = Publishes an event to the Server. INFO = Sends mid session information. REFER = Asks the recipient to issue call transfer. MESSAGE = Transports Instant Messages. UPDATE = Modifies the state of a session. SIP Request
1xx = Informational responses, such as 180 (RINGINGg). 2xx = Success responses, such as 200 (OK). 3xx = Redirection responses. 4XX = Request failures. 5xx = Server errors. 6xx = Global failures. SIP RESPONSE
Version 2 bits – version of RTP being used Padding 1 bit – padding bit determines additional padding octets at the end not part of the payload Extension 1 bit – determines whether there is a header extension CSRC 4 bits - # of CSRC sources Markers 1 bit – use differs upon application Payload type 7 bits Sequence number 16 bits Timestamp 32 bits SSRC 32 bits – synchronization source CSRC list 0 -15 items 32 bits RTP Packet