Internet Working 15th lecture last but one

1
Internet Working15th lecture (last but one)

• Chair of Communication Systems
• Department of Applied Sciences
• University of Freiburg
• 2005

1 43
2
Internet Workingadministrational stuff

• Next Thursday
• preliminary discussion of network seminars of the
  professorship (three different seminars, see
  homepage for description
• inaugural lecture at the faculty
• Next Friday written examination holidays -))
• Grades in oral or written exams will be sent to
  the examinations office (an will be available
  there beginning of winter term)
• If you need a special printed paper please tell
  us, so we could prepare it it will be available
  at the secretaries of the computing department

2 43
3
Internet WorkingLast lectures network security

• Firewalls for protecting machines from outside or
  outgoing traffic
• do not secure traffic in transit but try to block
  certain kinds of traffic
• operate on different layers of the OSI protocol
  stack
• MAC, IP, TCP/UDP header filtering
• Connection tracking (SYN-ACK of TCP handshake,
  sessions, ...)
• Special masquerading firewalls
• But security might be levelled with overcomplex
  firewalls
• Traffic can be tunneled over higher level
  protocols (piggybacking IP packets in DNS)
• IP-over-WAP2.0 tunnel (student project at our
  chair of CS)

3 43
4
Internet WorkingLast lecture network security
real time protocols

• Firewalls are only part of a network security
  concept often combined with VPN to span private
  networks of firms/organizations over insecure
  internet
• Firewalls do not protect by itself, but could be
  extended for spam, virus filters ... (operating
  often as proxies on application level)
• Second part of last lecture introduced to real
  time services for video broadcasting,
  voice-over-IP, internet telephony
• We will introduce SIP session initialization
  protocol
• Telephony over IP networks
• Only session setup, but compression, packet
  transport left to other services like RTP and RTCP

4 43
5
Internet Working application layer protocols
internet telephony

• For a rather long time telephone and data
  networks were different entities remember the
  network taxonomy
• packet orientated vs. circuit switched
• packet orientation is rather efficient in
  bandwidth using but cannot give any guarantees on
  packet delivery
• bandwidth growth and optional QoS helped to offer
  service quality near to circuit switching
• Why to provide two completely different
  infrastructures for rather the same services?
• voice is just another piece of data (and not the
  biggest one compared to other applications and
  services in use)

5 43
6
Internet Working application layer protocols
internet telephony

• Voice-over-IP is a big hype at the moment
• every network equipment vendor has some products
  in its portfolio (even companies like Siemens are
  able to offer products conforming to standards!!)
• many new telephone companies evolve to offer
  services, the old providers have to think on new
  strategies
• all of them hope for reduce of costs and a source
  for roaring profits -)
• so TCP/IP is just used for another
  application/service
• this service has to meet some requirements

6 43
7
Internet Workinginternet telephony - requirements

• security
• reduced costs might induce new type of SPAM
  spit (spam over internet telephony)
• how to know that the caller is the one he claims
  to, same for the called partner
• compatibility to existing services
• routing of emergency calls
• location of emergency
• presence
• rebustness of servers and routes
• permanent updates of clients (mobile devices move
  from network to network)

7 43
8
Internet Workinginternet telephony - requirements

• Voice over IP should offer
• higher robustness (e.g. alternate routes)
• better voice quality
• mobility, multimedia and conferencing
• secure communication
• gateways to other telephone systems (GSM, UMTS,
  PSTN)
• 100 open standards
• hope of a combination of lower costs with better
  functionality

8 43
9
Internet Workinginternet telephony
infrastructure (idialized -))
9 43
10
Internet Workinginternet telephony - standards

• Requirements by VoIP services
• enough bandwidth for digitized audio stream (both
  directions!)
• minimal jitter and noise -gt later this lecture
• Two main VoIP standards
• H323 standard developed by Telcos - ITU (last
  lecture)
• SIP internet standard
• SIP is session initialization protocol
• developed by Henning Schulzrinne (Feb. 1999)
• IETF Standard RFC 2543 (March 1999)
• current RFC 3261 (June 2002)

10 43
11
Internet Workinginternet telephony - SIP

• SIP just for session setup not for transport of
  multimedia streams
• inspired by HTTP
• text based Peer-to-Peer application layer
  protocol
• using requests and replies to set up a connection

11 43
12
Internet Workinginternet telephony - SIP

• Requirements toward SIP
• localization of endpoints
• setup of connections
• exchange of media and presence information
• modification of sessions rerouting and
  cancelling of calls
• complete a session
• scalability (more than one session should be
  possible)
• SIP addresses designed same way as email
  addresses
• sip userID_at_sipgateway.site

12 43
13
Internet WorkingSIP - entities

• Peers User Agents (UA)
• a UA can fulfill on of the following roles
• user agent client (UAC) initiator of a
  request
• user agent server (UAS) application, which
  contacts the user and answers requests for him
• SIP clients
• telephones as UAC or UAS
• Gateways connections to other networks,
  translates between different audio and video
  codecs
• SIP server
• might act as proxy server and could be used for
• authentification, authorization
• secure routing and rerouting

13 43
14
Internet WorkingSIP server

• SIP server
• redirect server information service
• location server is the request address for the
  host on wich a given user might be reached on
• registrar server acts as registration service
• registers the current location of the clients
• often at the same place as proxy or redirect
• is not a required component for SIP, but useful
  in larger setups

14 43
15
Internet WorkingSIP message types

• SIP defines messages for communication setup end
  ending

15 43
16
Internet WorkingSIP direct example session

• direct SIP connection
• disadvantage
• the calling party has to know the IP address of
  called party
• INVITE message contains the details, which type
  of session is to be initiated

16 43
17
Internet WorkingSIP direct example session
17 43
18
Internet WorkingSIP header fields

• Request URI, SIP version number
• VIA SIP version number, protocol, every SIP
  entity adds host and port, which created or
  routed the message
• Max-Forwards is decremented at every hop
• To, From tags as identifier
• Call-ID sender creates local non-ambiguous
  identifier which is globally unique in
  combination with the full qualified domain name
• CSeq command sequence is incremented with every
  new request
• More optional fields
• Contact contains the SIP address of the current
  host, if connected over proxy messages could be
  sent directly
• Content-Type and Length tell the style of the
  following SDP body

18 43
19
Internet WorkingSIP trying message (message
before ringing)
19 43
20
Internet WorkingSIP ringing message
20 43
21
Internet WorkingSIP ringing (cont.)

• To and From fields are the same as in INVITE
• direction of the initiating request is important
• connection over a proxy
• only answers to requests, does not send requests
  by itself
• no media abilities (does not handle media
  sessions)
• reads header and does not analyse body
• proxy may send request for clients location to
  location server

21 43
22
Internet WorkingSIP OK (200) message
22 43
23
Internet WorkingSIP redirect, registering
instant messaging

• redirection
• client sends INVITE to the SIP redirect server
• redirect server sends a request to the location
  server or requests the IP of the client to call
• current data is sent to the client, which ACK's
• from now on further on like direct connection
• registration
• REGISTER message to SIP registration server
• binding of the SIP URI with IP the users
  client/machine
• 200 OK
• instant messaging like the wellknown tools in
  that sector
• online status, buddy lists ...

23 43
24
Internet WorkingSDP service dscription protocol

• session description protocol (SDP)
• IETF standard RFC 2327
• text coded like SIP
• description syntax

24 43
25
Internet WorkingSDP service dscription protocol

• example
• v0
• ocalling 2890844526 2890844526 IN IP4 10.8.4.254
• sPhone Call
• cIN IP4 100.101.102.103
• t0
• maudio 49170 RTP/AVP
• artpmap0 PCMU/8000
• Version is 0 (at the moment no other versions
  available)
• Origin ousername session-id version network-type
  adress-type adress
• Subject ssubject

25 43
26
Internet WorkingSDP service description
protocol (cont.)

• Connection Data cnetwork-type address-type
  connection-adress
• Time tstart-time stop-time
• Media Announcements mmedia port transport
  format-list
• Attributes a
• This setup defines the multimedia session
• which usually uses RTP / RTCP explained later
  this lecture

26 43
27
Internet WorkingSIP firewalls, NAT, ...

• NAT
• SIP messages contain IP addresses in the data
  segments of its packets
• internal network addresses from the NATted
  network are not visible from the outside world
• A calls B, B gets the message from A, but not
  vice versa
• problem could be solved with a proxy server
  sitting in the internal and external LAN
• Firewalls
• RTP does not use fixed layer 4 port numbers
• variable in the range of 1024 - 65534

27 43
28
Internet WorkingSIP firewalls, NAT, ... (cont.)

• stun protocol
• simple traversal of UDP through NATs
• returning public's IP port
• can help to determine which kind of NAT is used
• most clients implement that protocol to produce
  the relevant SDP messages
• stun server will send its response to the IPport
  the initial packet was sent to
• if change-ip flag, then sends from different IP
• if change-port flag from different port

28 43
29
Internet Workingreal time services

• introduced SIP
• does not handle multimedia streams but only
  session setup
• setup is rather uncritical, the multimedia stream
  (the phone call taking place) is not
• requirements toward networks for real-time audio
  and video at least
• short delay (delay is composed from several
  parameters) and enough bandwidth
• normally available in backbone networks
• But more problematic the the (private) end user
  over low bandwidth connections

29 43
30
Internet Workingreal time services

• During maturing of the internet bandwidth was
  often scarce and expensive
• many solutions to bandwidth management addressed
  the whole end-to-end system connection
• but most concepts (e.g. the ToS flag in IP
  header) are not really used
• By now It is often cheaper to add bandwidth than
  operating sophisticated bandwidth management
• But there are scenarios where quality of service
  (QoS) may improve the whole networks usability ...

30 43
31
Internet Workingrequirements towards network

• Voice over IP and Quality of Service
• Major challenges delay and delay variation
  (jitter)
• Delay jitter is the variability of
  source-to-destination delays of packets within
  the same packet stream
• Voice applications are usually interactive
• delay requirement for a telephone system
  150ms-250ms
• We identified the sources of delay in a voice
  over IP system
• OS delay 10s-100s milliseconds (digitisazion of
  data, compression and inter software data
  handling) ...

31 43
32
Internet Workingrequirements towards network

• Source jitter
• Network network conditions vary at different
  times.
• Non-real time OS samples processed at different
  time
• Jitter control - buffering at the destination
  task of the application used
• QoS parameters which should be taken into
  account
• Accuracy, latency
• Jitter and codec quality
• Talked on SIP after session establishment RTCP
  and RTP data streams
• Depending on codec used a data stream of e.g.
  80kbit/s is generated for each direction
  (64kbit/s of ISDN PCM plus IP and UDP header)

32 43
33
Internet WorkingReal Time Protocol (RTP)

• Introduction of a special multimedia protocol
• Video and audio streaming
• Defined in RFC 1889
• Used for transporting common formats such as PCM
  and GSM for sound , and MPEG1 and MPEG2 for video
• RTP can be viewed as a sublayer of the transport
  layer
• Usually on top of UDP
• 8byte header (faster transfer)
• No setup overhead like with TCP session
• no explicit connection handling (left to
  protocols like SIP) faster

33 43
34
Internet WorkingSIP benefits over other
protocols/solutions like H323

• RTP packet header
• Payload type (7 bits) the type of audio/video
  encoding
• Sequence number (16 bits)
• Time stamp (32 bits) use for jitter removal -
  derived from a sampling clock at the sender
• Synchronization Source Identifier (SSRC) (32
  bits) identify the source of the RTP stream
• It is not the IP address of the sender (would
  violate the layering) but a number that the
  source assigns randomly when the new stream is
  started

34 43
35
Internet Workingreal time protocol
35 43
36
Internet WorkingRTP

• At the sender, the application puts its
  audio/video data with an RTP header and sends
  into the UDP socket
• The application in the receiver extracts the
  audio/video data from the RTP packet
• Uses the header fields of the RTP packet to
  properly decode and playback the audio/video data
• Helper protocol RTCP (RTP Control Protocol)
• RTCP packets do not encapsulate audio/video data

36 43
37
Internet WorkingRTCP

• RTCP packets sent periodically between sender and
  receivers to gather useful statistics
• number of packets sent
• number of packets lost
• interarrival jitter
• RTP and RTCP packets are distinguished from each
  other through the use of distinct port numbers

37 43
38
Internet Workingreal time control protocol
38 43
39
Internet WorkingRSVP

• RTP needs a bandwidth at least of the rate as
  packets are sent in each direction
• Otherwise packet loss or delays will occur and
  decrease the quality of data stream
• A special protocol was developed to add service
  quality parameters to the packet orientated
  internet
• RSVP - part of a larger effort to enhance the
  current Internet architecture with support for
  Quality of Service flows
• RFC 2205
• RSVP requests will generally result in resources
  being reserved in each node along the data path
• Resource we speak of is bandwidth (delay is much
  more complicated to reserve within IP networks)

39 43
40
Internet WorkingRSVP

• Signaling protocol introduced to reserve
  bandwidth between a source and its corresponding
  destination
• Main features of RSVP are
• Use of soft state'' in the routers
• receiver-controlled reservation requests
• flexible control over sharing of reservations
• forwarding of subflows
• the use of IP multicast for data distribution
• Source ? Destination RSVP path message
• Destination ? Source RSVP reserve message
• Nice try but ...

40 43
41
Internet WorkingRSVP problems

• Routers cannot not store state information about
  packets often too slow
• Simpler technique mark each packet with a simple
  flag indicating how to treat it
• Individual flows are classified into different
  traffic classes
• Each router sorts packets into queues via 
  differentiated services (DS) flag
• Queues get different treatment (e.g. priority,
  share of bandwidth, probability of discard)

41 43
42
Internet WorkingRSVP problems

• Result is coarsely predictable class of service
  for each differenciated services field value
• Cost of transmission varies by type of service
• Each traffic class is reserved a defined level of
  resources, e.g. buffer and bandwidth
• Different QoS guarantee policies can be applied
  in different traffic classes
• When congestion occurs, packets in low priority
  traffic classes will be dropped first
• The buffer and the bandwidth in a router for high
  priority traffic classes are more than low
  priority traffic classes
• More scalable than RSVP but cannot allocate
  resources precisely

42 43
43
Internet Workingliterature

• SIP
• Kurose Ross Computer Networking, 3rd edition
  (international)
• Section 7.4.3 SIP
• Tanenbaum Computer Networks, 4th edition
• Section 7.4.5 Voice over IP

43 43