Transporting Voice by Using IP

Chapter 2
Internet Overview
„ A collection of networks
„ The private networks
„ LANs, WANs
„ Institutions, corporations, business and government
„ May use various communication protocols
„ The public networks
„ ISP: Internet Service Providers
„ Using Internet Protocol
„ To connect to the Internet
„ Using IP

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Interconnecting Networks

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 Overview of the IP Protocol Suite

„ IP
„ A routing protocol for the passing of data packets
„ Must work in cooperation with higher layer protocols
and lower-layer transmission systems
„ The OSI seven-layer model
„ The top layer: useable information to be passed to
the other side
„ The information must be
„ Packaged appropriately
„ Routed correctly
„ And it must traverse some physical medium

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OSI Model [1/3]

„ Physical layer
„ The physical media
„ Coding and modulation schemes for 1’s and 0’s
„ Data link layer
„ Transport the information over a single link
„ Frame packaging, error detection/correction and
retransmission
„ Network layer
„ Routing traffic through a network
„ Passing through intermediate points

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 OSI Model [2/3]

„ Transport layer
„ Ensure error-free, omission-free and in-sequence
delivery
„ Support multiple streams from the source to
destination for applications
„ Session layer
„ The commencement (e.g., login) and completion
(e.g., logout) of a session between applications
„ Establish the dialogue
„ One way at a time or both ways at the same time

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OSI Model [3/3]

„ Presentation layer
„ Specify the language, the encoding and so on
„ Application layer
„ Provide an interface to the user
„ File transfer programs and web browsers

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The IP suite and the OSI stack
„ TCP
„ Reliable, error-free, in-sequence delivery
„ UDP
„ No sequencing, no retransmission

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 Internet Standards and the Process

„ The Internet Society

„ A non-profit organization
„ Keep the Internet alive and growing
„ “To assure the open development, evolution, and
the use of Internet for the benefit of all people
throughout the world”
„ The tasks include
„ Supporting the development and dissemination of Internet
standards
„ Supporting the RD related to the Internet and
internetworking
„ Assisting developing countries

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Internet Standards and the Process
„ IAB
„ The Internet Architecture Board
„ The technical advisory group
„ Providing technical guidance to Internet Society
„ Overseeing the Internet standards process
„ IETF
„ The Internet Engineering Task Force
„ Comprising a huge number of volunteers
„ Equipment vendors, network operators, research institutions
etc.
„ Developing Internet standards
„ Detailed technical work
„ Working groups
„ megaco, iptel, sip, sigtran
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Internet Standards and the Process

„ IESG
„ The Internet Engineering Steering Group
„ Managing the IETF’s activities
„ Approving an official standard
„ IANA
„ The Internet Assigned Numbers Authority
„ Unique numbers and parameters used in Internet
standards
„ Be registered with the IANA

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The Internet Standards Process

„ The process
„ RFC 2026
„ First, Internet Draft
„ The early version of spec.
„ Can be updated, replaced, or made obsolete by
another document at any time
„ IETF’s Internet Drafts directory
„ Six-month life-time

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The Internet Standards Process

„ RFC
„ Request for Comments
„ An RFC number
„ Proposed standard
„ A stable, complete, and well-understood spec.
„ Has garnered significant interest
„ Draft standard
„ Two independently successful implementations
„ Interoperability be demonstrated

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The Internet Standards Process

„ A standard
„ The IESG is satisfied
„ The spec. is stable and mature
„ Significant operational experience
„ A standard (STD) number
„ Not all RFCs are standards
„ Some document Best Current Practices (BCPs)
„ Processes, policies, or operational considerations
„ Others applicability statements
„ How a spec be used, or different specs work together

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 IP

„ RFC 791
„ Amendments: RFCs 950, 919, and 920
„ Requirements for Internet hosts: RFCs 1122, 1123
„ Requirements for IP routers: RFC 1812
„ IP datagram
„ Data packet with an IP header
„ Best-effort protocol
„ No guarantee that a given packet will be delivered

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IP Header [1/2]
„ Version 4
„ Header Length
„ Type of Service
„ Total Length
„ Identification, Flags, and Fragment Offset
„ A datagram can be split into fragments
„ Identify data fragments
„ Flags
„ a datagram can be fragmented or not
„ Indicate the last fragment
„ TTL
„ A number of hops (not a number of seconds)

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IP Header [2/2]
„ Protocol
„ The higher-layer protocol
„ TCP (6); UDP (17)
„ Source and Destination IP Addresses

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IP Routing

„ Based on the destination address in the IP

header
„ Routers
„ Can contain a range of different interfaces
„ Determine the best outgoing interface for a
given IP datagram
„ Routing table
„ Destination
„ IP route mask
„ For example, any address starting with 182.16.16
should be routed on interface A. (IP route mask
255.255.255.0)

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Populating Routing Tables

„ Issues
„ The correct information in the first place
„ Keep the information current in a dynamic
environment
„ The best path?
„ Protocols
„ OSPF (Open Short Path First)
„ An AS (Autonomous System) is a group of routers that
share routing information between them.
„ Area 0: backbone area
„ Border router
„ BGP (Border Gateway Protocol)
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OSPF Areas

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 TCP

„ Transmission Control Protocol

„ In sequence, without omissions and errors
„ End-to-end confirmation, packet retransmission,
Flow control
„ RFC 793
„ Break up a data stream in segments
„ Attach a TCP header
„ Sent down the stack to IP
„ At the destination, checks the header for errors
„ Send back an ack
„ The source retransmits if no ack within a given
period
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The TCP Header [1/5]

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The TCP Header [2/5]

„ TCP Port Numbers

„ Identifying a specific instance of a given
application
„ A unique port number for a particular session
„ Well-known port numbers
„ IANA, 0-1023
„ 23, telnet; 25, SMTP
„ Many clients and a server
„ TCP/IP
„ Source address and port number + Destination address
and port number
„ A socket address (or a transport address)

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The TCP Header [3/5]
„ Sequence and acknowledge numbers
„ Identify individual segments
„ Actually count data octets transmitted
„ A given segment with a SN of 100 and contains 150 octets
of data
„ The ack number will be 250
„ The SN of the next segment is 250
„ Other header fields
„ Data offset: header length (in 32-bit words)
„ URG: 1 if urgent data is included, use urgent pointer field
„ ACK: 1, an ACK
„ PSH: a push function, be delivered promptly

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The TCP Header [4/5]

„ RST: reset; an error and abort a session

„ SYN: Synchronize; the initial messages
„ FIN: Finish; close a session
„ Window
„ The amount of buffer space available for receiving
data
„ Checksum

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The TCP Header [5/5]

„ Urgent Pointer
„ An offset to the first segment after the urgent
data
„ Indicates the length of the urgent data
„ Critical information to be sent to the user
application ASAP

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 TCP Connections

„ An example
„ After receiving
„ 100, 200, 300
„ ACK 400
„ Closing a connection
„ Æ FIN
„ Å ACK, FIN
„ Æ ACK

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UDP
„ User Datagram Protocol
„ Pass individual pieces of data from an application to IP
„ No ACK, inherently unreliable
„ Applications
„ A quick, on-shot transmission of data, request/response
„ DNS
„ If no response, the AP retransmits the request
„ The AP includes a request identifier
„ The source port number is optional
„ Checksum

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Voice over UDP, not TCP

„ Speech
„ Small packets, 10 – 40 ms
„ Occasional packet loss is not a catastrophe
„ Delay-sensitive
„ TCP: connection set-up, ack, retransmit → delays
„ 5 % packet loss is acceptable if evenly spaced
„ Resource management and reservation techniques
„ A managed IP network
„ In-sequence delivery
„ Mostly yes
„ UDP was not designed for voice traffic

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 The Real-Time Transport Protocol

„ RTP: A Transport Protocol for Real-Time

Applications
„ RFC 1889
„ RTP – Real-Time Transport Protocol
„ RTCP – RTP Control Protocol
„ UDP
„ Packets may be lost or out-of-sequence
„ RTP over UDP
„ A sequence number
„ A time stamp for synchronized play-out
„ Does not solve the problems; simply provides
additional information
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 RTCP

„ A companion protocol
„ Exchange messages between session users
„ # of lost packets, delay and inter-arrival jitter
„ Quality feedback
„ RTCP is implicitly open when an RTP session
is open
„ E.g., RTP/RTCP uses UDP port 5004/5005

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 RTP Payload Formats [1/2]

„ RTP carries the actual digitally encoded voice

„ RTP header + a payload of voice/video samples
„ UDP and IP headers are attached
„ Many voice- and video-coding standards
„ A payload type identifier in the RTP header
„ Specified in RFC 1890
„ New coding schemes have become available
„ See table 2-1
„ A sender has no idea what coding schemes a
receiver could handle

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RTP Payload Formats [2/2]

„ Separate signaling systems

„ Capability negotiation during the call setup
„ SIP and SDP
„ A dynamic payload type may be used
„ Support new coding scheme in the future
„ The encoding name is also significant.
„ Unambiguously refer to a particular payload specification
„ Should be registered with the IANA

„ RED, Redundant payload type

„ Voice samples + previous samples
„ May use different encoding schemes
„ Cope with packet loss
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RTP Header Format

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The RTP Header [1/4]

„ Version (V)
„ 2
„ Padding (P)
„ The padding octets at the end of the payload
„ The payload needs to align with 32-bit boundary
„ The last octet of the payload contains a count of
the padding octets.
„ Extension (X)
„ 1, contains a header extension

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 The RTP Header [2/4]

„ CSRC Count (CC)

„ The number of contributing source identifiers
„ Marker (M)
„ Support silence suppression
„ The first packet of a talkspurt, after a silence period
„ Payload Type (PT)
„ In general, a single RTP packet will contain media coded
according to only one payload format.
„ RED is an exception.
„ Sequence number
„ A random number generated by the sender at the beginning
of a session
„ Incremented by one for each RTP packet

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 The RTP Header [3/4]

„ Timestamp
„ 32-bit
„ The instant at which the first sample
„ The receiver
„ Synchronized play-out
„ Calculate the jitter
„ The clock freq depends on the encoding
„ E.g., 8000Hz
„ Support silence suppression
„ The initial timestamp is a random number chosen by the
sending application.

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The RTP Header [4/4]
„ Synchronization Source (SSRC)
„ 32-bit identifier
„ The entity setting the sequence number and timestamp
„ Chosen randomly, independent of the network address
„ Meant to be globally unique within a session
„ May be a sender or a mixer
„ Contributing Source (CSRC)
„ An SSRC value for a contributor
„ 0-15 CSRC entries
„ RTP Header Extensions

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Mixers and Translators
„ Mixers
„ Enable multiple media
streams from different
sources to be combined into
a single stream
„ If the capacity or
bandwidth of a participant
is limited
„ An audio conference
„ The SSRC is the mixer
„ More than one CSRC values
„ Translators
„ Manage communications
between entities that does
not support the same
coding scheme
„ The SSRC is the participant,
not the translator.

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The RTP Control Protocol [1/3]

„ RTCP
„ A companion control protocol of RTP
„ Periodic exchange of control information
„ For quality-related feedback
„ A third party can also monitor session quality and
detect network problems.
„ Using RTCP and IP multicast
„ Five types of RTCP packets
„ Sender Report: transmission and reception statistics
„ Receiver Report: reception statistics

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The RTP Control Protocol [2/3]

„ Source Description (SDES)

„ One or more descriptions related to a particular session
participant
„ Must contain a canonical name (CNAME)
„ Separate from SSRC which might change
„ When both audio and video streams were being
transmitted, the two streams would have
„ different SSRCs

„ the same CNAME for synchronized play-out

„ BYE
„ The end of a participation in a session
„ APP
„ For application-specific functions

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The RTP Control Protocol [3/3]
„ Two or more RTCP packets will be combined
„ SRs and RRs should be sent as often as possible to allow
better statistical resolution.
„ New receivers in a session must receive CNAME very
quickly to allow a correlation between media sources and
the received media.
„ Every RTCP packet must contain a report packet (SR/RR)
and an SDES packet
„ Even if no data to report
„ An example RTP compound packet

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 RTCP Sender Report

„ SR
„ Header Info
„ Sender Info
„ Receiver Report Blocks
„ Option
„ Profile-specific extension

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Header Info

„ Resemble to an RTP packet

„ Version
„ 2
„ Padding bit
„ Padding octets?
„ RC, report count
„ The number of reception report blocks
„ 5-bit
„ If more than 31 reports, an RR is added
„ PT, payload type

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Sender Info
„ SSRC of sender
„ NTP Timestamp
„ Network Time Protocol Timestamp
„ The time elapsed in seconds since 00:00, 1/1/1900 (GMT)
„ 64-bit
„ 32 MSB: the number of seconds
„ 32 LSB: the fraction of a seconds (200 ps)
„ RTP Timestamp
„ Corresponding to the NTP timestamp
„ The same as used for RTP timestamps
„ For better synchronization
„ Sender’s packet count
„ Cumulative within a session
„ Sender’s octet count
„ Cumulative within a session
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 RR blocks [1/2]

„ SSRC_n
„ The source identifier of the session participant to which the
data in this RR block pertains.
„ Fraction lost
„ Fraction of packets lost since the last report issued by this
participant
„ By examining the sequence numbers in the RTP header
„ Cumulative number of packets lost
„ Since the beginning of the RTP session
„ Extended highest sequence number received
„ The sequence number of the last RTP packet received
„ 16 lsb, the last sequence number
„ 16 msb, the number of sequence number cycles

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 RR blocks [2/2]

„ Interarrival jitter
„ An estimate of the variance in RTP packet arrival
„ Last SR Timestamp (LSR)
„ The middle 32 bits of the NTP timestamp used in
the last SR received from the source in question
„ Used to check if the last SR has been received
„ Delay Since Last SR (DLSR)
„ The duration in units of 1/65,536 seconds

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 RTCP Receiver Report

„ RR
„ Issued by a participant who receives RTP packets
but does not send, or has not yet sent
„ Is almost identical to an SR
„ PT = 201
„ No sender information

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RTCP Source Description Packet

„ Provides identification and information

regarding session participants
„ Must exist in every RTCP compound packet
„ Header
„ V, P, SC, PT=202, Length
„ Zero or more chunks of information
„ An SSRC or CSRC value
„ One or more identifiers and pieces of information
„ A unique CNAME
„ Email address, phone number, name

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„ RTCP BYE Packet
„ Indicate one or more media sources are no longer
active
„ Application-Defined RTCP Packet
„ For application-specific data
„ For non-standardized application

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 Calculating Round-Trip Time

„ Use SRs and RRs

„ E.g.
„ Report A: A, T1 → B, T2 A B
„ Report B: B, T3 → A, T4
T1
„ RTT = T4-T3+T2-T1
„ RTT = T4-(T3-T2)-T1 T2
„ Report B T3
„ LSR = T1
T4
„ DLSR = T3-T2

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 Calculation Jitter

„ The mean deviation of the difference in

packet spacing at the receiver
„ Si = the RTP timestamp for packet i
„ Ri = the time of arrival
„ D(i,j) = (Rj-Sj) - (Ri- Si)
„ The Jitter is calculated continuously
„ J(i) = J(i-1) + (| D(i-1,i) | - J(i-1))/16

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Timing of RTCP Packets
„ RTCP provides useful feedback
„ Regarding the quality of an RTP session
„ Delay, jitter, packet loss
„ Be sent as often as possible
„ Consume the bandwidth
„ Should be fixed at 5%
„ An algorithm, RFC 1889
„ Senders are collectively allowed at least 25% of
the control traffic bandwidth.
„ The interval > 5 seconds
„ 0.5 – 1.5 times the calculated interval
„ A dynamic estimate the avg. RTCP packet size

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IP Multicast

„ An IP diagram sent to multiple hosts

„ Conference
„ To a single address associated with all listeners
„ Multicast groups
„ Multicast address
„ Join a multicast group
„ Inform local routers
„ Routing protocols
„ Support propagation of routing information for multicast
addresses
„ Minimize the number of datagrams sent

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