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Introduction to VoIP Part II Dr. Farid Farahmand CET479 Updated 5/18/2007.

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Presentation on theme: "Introduction to VoIP Part II Dr. Farid Farahmand CET479 Updated 5/18/2007."— Presentation transcript:

1 Introduction to VoIP Part II Dr. Farid Farahmand CET479 Updated 5/18/2007

2 Overview Basic concepts of VoIP and its motivating facts How to digitally decode voice prior to its transport How to transport voice between users  After the session is established how to transport voice How to setup and teardown voice sessions  How to create sessions  How signaling protocols work

3 Speech Coding Voice has to be digitally encoded/decoded  Streams of 1’s and 0’s How voice is coded impacts the channel efficiency (BW utilization)  Various speech coding techniques are used Bandwidth and voice quality are related  Yet the relation is not linear  For example: 16 Kbps voice transmission is not necessarily better than 32 Kbps Objective of speech coding is to minimize BW and maintaining high quality of speech  High quality is measured by MOS metric (Mean-Option Score)  Other metric alternatives are available (PSQM)

4 A Little about Speech Speech is considered to be an analog signals  The objective is to reconstruct the speech digitally A signal can be reconstructed if the sampling rate is twice the max. input frequency More bits requires more BW but typically more quantization level

5 A Little about Speech Uniform quantization level can cause discrimination  Loud voices will have lower quantization error A more effective approach is to us non-uniform quantization  Smaller levels  smaller quantization level  Larger levels  Less granularity More accuracy Less accuracy

6 Speech-Coding Techniques Choice of speech coding is critical to having high-quality voice Two conflicting objectives  Reducing bandwidth  Maintaining the natural-sounding speech (toll quality)

7 G. 711 Speech Coding ITU Recommendation G. 711 Speech decoding  Typical human speech has a maximum frequency of about 4 KHz: F max = 4KHz  Based on Nyquist Theorem, analog signals must be sampled at twice their maximum frequency: Sampling rate =8000 sample/second = 2 x F max  Each sample is represented with 8 bits  BW requirement will be 64 Kbps for standard circuit switch based telephone  Toll-quality (MOS) is 4.3 = Excellenet More efficient coding techniques  G.726  32 Bit rate (Kbps) toll-quality = 4.0  G.728  16 Bit rate (Kbps) toll-quality = 3.9  G.729  08 Bit rate (Kbps) toll-quality = 4.0 VoIP uses more efficient coding techniques  The two ends negotiate on which coding technique to use

8 Next: Basic concepts of VoIP and its motivating facts How to digitally decode voice prior to its transport How to transport voice between users  After the session is established how to transport voice How to setup and teardown voice sessions  How to create sessions  How signaling protocols work

9 Transporting Voice Signals Digitally codes voice can be encapsulated into IP packets  IP is just a routing protocol  IP routing is based on the destination address – packets with the same source/destination address can take different paths  Provides no quarantine of service One way to transport the IP packet packets is using TCP The transmission control protocol (TCP)  Ensuring that all packet are delivered in sequence  Providing transmission reliability  TCP provides port number in its header to distinguish between different applications (SMTP: Port 25 / Web: port 80 / Telnet: Port 23)

10 TCP/IP Model (Click for more information) The five layer TCP/IP model 5. Application layerApplication layer DHCPDHCP DNS FTP HTTP IMAP4 IRC NNTP MIME POP3 SIP SMTP SNMP SSH TELNET BGP RPC RTP RTCP TLS/SSL SDP SOAP L2TP PPTP …DNSFTPHTTPIMAP4IRCNNTPMIMEPOP3SIPSMTP SNMPSSHTELNETBGPRPCRTPRTCPTLS/SSLSDPSOAP L2TPPPTP 4. Transport layerTransport layer TCPTCP UDP DCCP SCTP GTP …UDPDCCPSCTPGTP 3. Network layerNetwork layer IPIP (IPv4 IPv6) ARP RARP ICMP IGMP RSVP IPSec …IPv4IPv6ARPRARPICMPIGMPRSVPIPSec 2. Data link layerData link layer ATMATM DTM Ethernet FDDI Frame Relay GPRS PPP …DTMEthernetFDDIFrame RelayGPRSPPP 1. Physical layerPhysical layer Ethernet physical layerEthernet physical layer ISDN Modems PLC RS232 SONET/SDH G.709 Wi-Fi …ISDNModemsPLCRS232SONET/SDHG.709 Wi-Fi

11 TCP/IP Headers

12 Introduction to UDP The User Defined Protocol performs a very simple function  Passing IP packets to the end user  Provides no guarantee of service and inherently unreliable  Has no concept of packet ordering  Yet, provides a quick one-shot transmission  Most common example is using UDP in DNS

13 UDP Field Name Size (byte s) Description Source Port2 Source Port: The 16-bit port number of the process that originated the UDP message on the source device. This will normally be an ephemeral (client) port number for a request sent by a client to a server, or a well-known/registered (server) port number for a reply sent by a server to a client. See the section describing port numbers for details.serverSee the section describing port numbers for details Destination Port 2 Destination Port: The 16-bit port number of the process that is the ultimate intended recipient of the message on the destination device. This will usually be a well- known/registered (server) port number for a client request, or an ephemeral (client) port number for a server reply. Again, see the section describing port numbers for details.see the section describing port numbers for details Length2 Length: The length of the entire UDP datagram, including both header and Data fields. Checksum2 Checksum: An optional 16-bit checksum computed over the entire UDP datagram plus a special “pseudo header” of fields. See below for more information. DataVariableData: The encapsulated higher-layer message to be sent.

14 Voice over UDP UDP was not designed for transporting voice Due to its quick transporting ability, it is suitable for voice Basic shortcoming of UDP  No packet loss recovery mechanism Voice communications can tolerate some loss Efficient coding techniques can be design to recover some lost packets Supporting QoS can reduce the probability of packet loss  No packet ordering scheme Packets in the same session are unlikely to follow different paths  lower probability of out of ordering …we still like to resolve some of the shortcomings of UDP

15 A Transport Protocol for Real-Time Application Protocol (RTP) RTP is designed to support transporting real-time applications (voice, video, etc.) RTP contains two protocols  RTP  RTP Control Protocol Main functionalities  Detect packet out-of- sequencing  Report packet loss  Only provides information and takes no action!

16 RTP Protocols RTP resides on top of UDP  Includes packet sequence number  Provides timestamp (used for synchronization and calculating jitter and delay) RTP Control Protocol (RTCP)  Considered as a companion to RTP / optional  Provides feedback about quality of the voice session Number of lost RTP packets Packet delays Inter-arrival jitter RTP and RTCP are often established as two separate sessions  Odd/Even port numbers between ,535

17 Next: Basic concepts of VoIP and its motivating facts How to digitally decode voice prior to its transport How to transport voice between users  After the session is established how to transport voice How to setup and teardown voice sessions  How to create sessions  How signaling protocols work

18 Call Setup and Teardown The main question:  How to establish a voice session  How to teardown the session Call setup and teardown is commonly used in traditional telephony Signaling protocols are invoked before and during the call  Setup  Monitor/maintenance  Teardown SS7 is the most common signaling example used in our telephone network In case of VoIP most initial signaling protocols were proprietary ITU-T (International Telecommunications Union Telecommunications Standardization Sector) recommended H.323 as the signaling protocol  Version 1: 1996  Version 2: 1998  Version 4: Today!

19 H.323 Architecture Basic components and scope  Terminal Endpoints / end-user communication devices  Multipoint control unit (MCU) An H.323 endpoint supporting multipoint conference  Gatekeeper Optional entity Controls a number of H.323 terminal, gateways and MCUs Offers BW control services used to support QoS  Gateway Establishes connection to other networks (etc. ISDN) Provides translation services between H.323 and other types of networks A set of terminals, MCUs, that a single gatekeeper controls is called a ZONE SCN = traditional switched circuit network (SCN)

20 General Idea

21 Overview of H.323 Protocols The actual signaling messages between H.323 entities are specified by  H.225 RAS Signaling  H.223 Call Signaling  H.245 Control Signaling H.225 has two parts  Call Signaling: The setup and teardown signaling is very similar to ISDN layer 3 spec. (Q.931) Can be carried over UDP or TCP / can be performed together – whichever is established first  RAS (registration, admission and Status) signaling Used between endpoint and a gatekeeper Always carried over UDP

22 Overview of H.323 Protocols H.245 is a control protocol used between two or more endpoints  Manages the media streams between H.323 session participants  Establishes logical channels between endpoints The channel carries media streams between participants and include media type, bit rate, and so on

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26 References es/40-04/blackfin_voip.html es/40-04/blackfin_voip.html - RTCP


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