Advanced Communication Networks

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1 Advanced Communication Networks Advanced Transport Issues Prof. Ana Aguiar University of Porto, FEUP

2 Contents Congestion in Best-effort Networks TCP Congestion Control Congestion Avoidance Mechanisms Streaming and Real-Time Transport Protocols RTP, RTCP, RTSP, RoHC Recent Transport Protocols SCTP, DCCP

3 Network Model Host Router Host Host Router Router Host Router Router Host Host Router Router Router Host Host

4 Best Effort Service Model IP networks typically offer same treatment of all packets and no service guarantees Packets are forwarded if and when possible Packets are considered independent/ unrelated Matters pertaining to flows are typically handled by the hosts at higher layers TCP, RTP/ RTCP,

5 Best-effort Networks Best-effort networks treat all packets alike and network does not talk to hosts Sometimes congestion happens Need to react to or avoid congestion Congestion control and avoidance mechanisms

6 Dealing with Congestion Congestion control: reacting to congestion Host based: TCP congestion control Based on detecting congestion Congestion avoidance: preventing congestion Active Queue Management is host and router based: routers actively signal expected congestion Can also be only host based: hosts predict congestion based on RTT observation

7 TCP Congestion Control End-to-end mechanism that enables each source to adapt send rate according to available bandwidth Deals with congestion and variable bandwidth Window-based and feedback-based Different from flow control!

8 TCP Flow Control Receiver advertises his capability in the form of a window size Advertised window size is used to calculate effective window size at any time EffectiveWindow = AdvertisedWindow (LastByteSent - LastByteACKed)

9 TCP Congestion Control Change calculation of effective window size to accommodate congestion control window MaxWindow = min(congestionwindow, AdvertisedWindow) EffectiveWindow = MaxWindow (LastByteSent LastByteACKed)

10 TCP Congestion Control Three mechanisms control the congestion control window Additive increase/ multiplicative decrease Slow start Fast retransmit/ fast recovery Goal: use the available bandwidth as much as possible without causing buffer overflow at the routers

11 Estimating Congestion Implemented at the hosts without cooperation/ support of the (best-effort) network How can a host estimate available bandwidth? In fixed networks, there are almost no packet losses due to errors TCP assumes lost bytes are due to congestion Dropped at routers Arrived too late (timeout)

12 Additive Increase/ Multiplicative Decrease Mechanism to adapt congestion window to available bandwidth Lost packet => congestion: decrease congestion window by half Successful packet => less congestions: increase congestion window Increment = MSS x (MSS/CongestionWindow) MSS : Maximum Segment Size CongestionWindow += Increment

13 Additive Increase/ Multiplicative Decrease Evolution of CongestionWindow with time for a connection in steady-state follows a sawtooth pattern And so does the throughput Implicit fairness among TCP flows Image in Computer Networks: a Systems Approach, L. Peterson, B. Davie

14 Additive Increase/ Multiplicative Decrease Why decrease faster than increase? Consequences of congestion are more severe than consequences of not fully utilising link Has been proven through analysis that multiplicative decrease guarantees stability, i.e. guarantees that queues will reduce size in acceptable time

15 Slow Start Mechanism to discover available bandwidth during initial phase of a connection Goal is not to flood the network with a burst of packets equivalent to a full transmission window

16 Slow Start CongestionWindow = 1 For every successfully sent packet CongestionWindow *= 2 Congestion window grows exponentially ACK ACK ACK ACK Data ACK Data Data ACK ACK Data Data Data Data Data Data Data Data Data Data

17 Slow Start Also used after connection stops sending while waiting for an acknowledgement For example, when first packet in the window is lost Different scenario from connection start Value of CongestionWindow after loss is stored in CongestionThreshold CongestionWindow increases exponentially up to CongestionThreshold (slow start), and then additively

18 Fast Retransmit/ Fast Recovery Round Trip Time (RTT) calculation is coarse When a packet is lost, waiting for timeout before retransmission can be very ineffective When a single packet is lost, it should not be interpreted as congestion, but as a loss Sender sees one lost packet followed by s u c c e s s f u l o n e s a s duplicate acknowledgements

19 Fast Retransmit/ Fast Recovery Fast Retransmit : trigger retransmission after three d u p l i c a t e acknowledgements Packet after acknowledged byte is re-transmitted Fast Recovery: set CongestionWindow to half after a Fast Retransmit, instead of doing slow start as described previously ACK 1 ACK 2 ACK 2 ACK 2 ACK 2 ACK 2 ACK 2 ACK 7

20 Active Queue Management Mechanisms that detect congestion in advance and inform hosts to reduce send rate Congestion avoidance mechanisms that involve routers as well as hosts DECbit Random Early Detection (RED) Explicit Congestion Notification (ECN)

21 DECbit Bit in packet header can be set by routers with queues larger than threshold Receiving host will signal it back to sender Sender has a congestion window which is managed according to additive increase/ multiplicative decrease Increase window by 1 when less than 50% packets within the window were dropped Decrease window to 87.5% if more than 50% packets within the window are dropped

22 Random Early Detection (RED) Designed to be used with TCP Routers drop packets before congestion occurs to signal congestion to the source Source will react by activating congestion control mechanisms, reducing their send rate

23 Random Early Detection (RED) Router drops packets when average queue length exceeds a drop threshold Queue length monitored using a moving average AvgQLen = (1-α) x AvgQLen + α x SampleQLen 0 < α < 1 Senders react to drops not faster than one RTT ( 100ms) Image in Computer Networks: a Systems Approach, L. Peterson, B. Davie

24 Random Early Detection (RED) Router drops packets with probability P if AvgQLen is between MinThreshold and MaxThreshold Algorithm avoids burst drops by calculating P as an increasing function of enqueued packets since the last drop Router drops all arriving packets if AvgQLen is larger than MaxThreshold Images in Computer Networks: a Systems Approach, L. Peterson, B. Davie

25 Random Early Detection (RED) By randomly dropping packets, flows that use a larger bandwidth share have larger probability of seeing their packets dropped Simply because they send more packets So, RED is inherently fair in this sense Still, it is a best-effort approach, which treats all flows equally

26 Explicit Congestion Notification (ECN) Alternative to RED which uses IP header bits to signal congestion instead of dropping packets Congestion is signalled to receiver, but data arrives Receiver sets congestion bit in ACK Sender behaves as if the packet was dropped Does not cause data loss Uses 2 bits in TOS field of IP header 1 for the sender to signal that it supports ECN 1 for a router to signal congestion

27 Explicit Congestion Notification (ECN) Uses 2 bits in TOS field of IP header 1 for the sender to signal that it supports ECN 1 for a router to signal congestion Congestion is signalled to receiver without data loss Router sets congestion bit Receiver sets congestion bit in ACK Sender behaves as if the packet was dropped

28 Host-Based Congestion Avoidance Senders use observable parameters to detect that congestion will happen soon Increasing RTT of successive packets Combination of RTT and variations CongestionWindow Flattened sending rate or throughput rate, e. g. estimated as bytes in the network / RTT Senders adapt send rate to detected congestion

29 Transport for Real-time Traffic

30 Multimedia Streaming A/D Coding Compression Packetisation Capture Users interact with each other with the stream Strong time correlation between sender and receiver media Internet Render D/A Decoding De- Compression Playout Buffer

31 Streaming Issues Strict delay dependencies between sender and receiver Delay requirements depend on type of traffic Time correlation between different streams Interactivity with stream source Source and receiver must agree on applicationspecific details

32 Playout Buffer Buffer on the receiver side where frames are put after reassembly and before being played Compensates for delay variations Usually, application waits for the buffer to fill up to a certain threshold before starting to play, inducing a playout delay Relevant for live streaming, like Internet radio, football Decoder reads from buffer Frame to be played at time n must arrive before that, as well as all other frames that frame n depends on

33 Playout Buffer Nth frame should be played at time tn=t0+n/r t0: initial playout delay R: frame rate Packet not on time may imply frame not on time What can cause the playout buffer to be empty? Bandwidth shortage Long delay Too large delay variations

34 Multimedia Support Multimedia Traffic H i g h l y v a r i a b l e t r a f f i c characteristics and QoS requirements Strict timing requirements (QoS, interactivity, synch) Traffic specific agreement needed between sender and receiver Network Varying delivered bandwidth No delay guarantees Delay variations Traffic agnostic ENABLE and ADAPT!

35 Transport and Signalling What support does data from such applications need? What must sender and receiver agree on before, during and after engaging in communication What information must sender and receiver exchange during communication?

36 IETF Protocol Suite Obsolete In Internet Telephony: Architecture and Protocols -- an IETF Perspective, H. Schulzrinne, J. Rosenberg, IEEE Network, 1999

37 RTP (Real Time Protocol) Provides mechanisms for streaming data end-toend in real-time among multiple clients Flexible (media formats, transport protocols) Out-of-band singalling data plane: RTP control plane: RTCP

38 RTP (Real Time Protocol) Provides timing information Synchronisation of different media (lip synchronisation) Time-stamping (useful for playout buffer) Provides sequence numbers Identify losses, re-order packets Identifies source of transported media streams useful for conferencing Enables multiplexing different content

39 RTP Control Plane: RTCP Inform of network issues, e.g. congestion information Negotiate and adapt media specific characteristics Coding, compression, dynamic frame boundaries

40 RTP Header Padding CSRC Count Payload Type Sequence Nr Version Extension Marker SSRC: Synchronisation source identifier CSRC: Contributing sources identifier CSRC Count: number contributing sources

41 RTP Entities Transport address: combination of IP and port (endpoint) RTP media type: collection of payload types within a session RTP session: communication between a pair of transport addresses RTP multimedia session: a set of sessions among a group of participants Synchronisation Source (SSRC): source of a synchronised RTP stream Contribution Source (CSRC): source of stream contributing to combined stream produced by mixer

42 RTP Mixer Intermediate system receives and combines RTP packets from different sources Needs to establish synchronisation relationship between mixed streams Resulting stream goes to a new multicast address

43 RTP Translator Different participants in a multimedia session may use different data standards (formats, compression,...) Intermediate systems that does not change SSRC Can change payloads: transcode, change compression

44 RTP Media Profiles Header fields interpretation defined by media profile Payload type: identifies format and interpretation of payload E.g., RTP AVP: Audio/ Video Profile with minimal session control (RFC 3551) It provides interpretations of generic fields within the RTP specification suitable for audio and video conferences. In particular, this document defines a set of default mappings from payload type numbers to encodings. Marker: interpretation is defined by profile, e.g. frame boundaries Extension headers: may be defined to carry additional information, e.g. MPEG-4 transport streams

45 RTP offers protocol mechanisms to signal relevant information between sender and receiver Meaning of fields can be defined anew by each new media profile Separation of protocol from meaning of information!

46 IETF Protocol Suite Obsolete In Internet Telephony: Architecture and Protocols -- an IETF Perspective, H. Schulzrinne, J. Rosenberg, IEEE Network, 1999

47 Real Time Control Protocol (RTCP) Functionalities: feedback receiver quality provide means for adaptation: flow control, adaptive media coding RTCP provides periodic feedback to all members of an RTP session using the same distribution as media data reception statistics on quality (loss, delay, jitter) faults for network problem diagnosis distribution properties How is de-multiplexing of RTP and RTCP achieved?

48 RTCP Packets SR RR Sender report, for transmission and reception statistics from participants that are active senders Receiver report, for reception statistics from participants that are not active senders SDES Source description items, including CNAME BYE APP Indicates end of participation Application specific functions

49 RTCP Sender Reports NTP timestamp RTS timestamp Sender total packet count Sender total octet count Receiver reports of receivers

50 RTCP Receiver Reports Receivers report to sources Fraction lost blocks since last report Total number of lost packets Highest sequence number received Jitter estimate Time of last report Time since last report (These reports are re-distributed by the source)

51 RTCP Source Descriptor Contains several descriptors Each descriptor contains Source ID <Tag, length, value> Name Phone number Location

52 RTCP Performance Issues Signalling protocol: overhead vs performance trade-off Too many reports => occupies too much bandwidth Too little reports => low reaction capability RFC suggests RTCP bandwidth limitation to 5% 1.25% for sources 3.75% for receivers Increasing amount of receivers => less RR / receiver De-synchronise RTCP packets from different session participants If synchronised, RTCP packet bursts can occur

53 RTP Performance Issues Calculate the overhead for a G.729 voice over IP packet over IPv4 and IPv6.

54 Robust Header Compression Large per packet overhead RTP + UDP + IPv4 headers = = 40 Bytes Several header fields are constant or incremental Especially critical for bandwidth-limited systems Wireless! (GPRS, 3G, WLAN,...) See RFC 3096, Requirements for robust IP/UDP/RTP header compression, M. Degermark, (Jul 2001)

55 RoHC RObust Header Compression (ROHC): Framework and four profiles: RTP, UDP, ESP, and uncompressed, RFC 3095, Jul 2001 Targets mainly wireless cellular links

56 RoHC Header Field Classification Inferred: fields whose value can be inferred from other values of other fields Static: fields expected to be constant during stream's lifetime Static-def: fields whose values define the stream Static-known: fields expected to have well-known values (do not need to be communicated) Changing

57 Header Fields version hlen TOS Byte Total length v= 2 IP id flags Fragment offset TTL protocol header checksum source IP address destination IP address source port number destination port number UDP length UDP checksum P X CC X PT RTP seq number RTS timestamp SSRC identifiers CSRC identifiers IPv4 UDP RTP

58 ROHC Potential Savings IPv4 (Bytes) UDP (Bytes) RTP (Bytes) Inferred Total (Bytes) Static bits 0 2 bits bits Static-def Staticknown bits 0 2 bits bits Per packet: 28.4 Bytes!

59 ROHC State Machines ROHC is an interaction of one compressor and one decompressor state machines (SM) One pair of state machines per ROHC context Each state machine has 3 states with similar but not equal meaning Machines start in lowest compression state and transit upwards

60 ROHC Compressor SM Initialisation and refresh First order state Second order state Initialisation and refresh Initialise static parts of context or recover after failure First order Efficiently communicate irregularities in the stream Second order Header completely predictable; optimal compression

61 ROHC Decompressor SM No context Static context Full context No context Not yet successfully decompressed a packet Full context Normal mode of operation Static context Low overhead fallback for decompression failures

62 ROHC Operation Modes Uni-directional (U-mode) Initial mode; requires no feedback Bi-directional optimistic mode (O-mode) Feedback channel used to send update requests Bi-directional reliable mode (R-mode) Feedback channel used to acknowledge all context updates

63 IETF Protocol Suite Obsolete In Internet Telephony: Architecture and Protocols -- an IETF Perspective, H. Schulzrinne, J. Rosenberg, IEEE Network, 1999

64 RTSP RFC 2326: Real Time Stream Control Protocol The Real-Time Streaming Protocol (RTSP) establishes and controls either a single or several time-synchronized streams of continuous media such as audio and video. It does not typically deliver the continuous streams itself, although interleaving of the continuous media stream with the control stream is possible (see Section 10.12). In other words, RTSP acts as a "network remote control" for multimedia servers., In RFC2326, Schulzrinne et al, 1998

65 RTSP Features Supports: Retrieval of media from a server Invitation of a media server to a conference Recording of a conference Similar to HTTP, but with differences: Stateful Both client and server can issue requests Data carried by another protocol Independent of description protocol (SDP, SMIL, ) Works with unicast and multicast

66 RTSP Methods OPTIONS SETUP PLAY RECORD REDIRECT PAUSE TEARDOWN ANNOUNCE DESCRIBE Get available methods Establish transport Start playback Start recording Redirect client to new server Halt delivery but keep state Remove state Change description of media object Get description of media object SET_PARAMETER Device or encoding control

67 How are they related? HTTP GET Session Description Web Server RTSP SETUP Client RTSP PLAY RTP Audio RTP Video RTCP RTSP Pause RTSP Close Media Server

68 New Transport Protocols More recently, driven by the growth of Internet voice and video traffic, a new transport protocols have been proposed. SCTP: Stream Control Transport Protocol DCCP: Datagram Congestion Control Protocol

69 Limitations of TCP Handshake Acknowledgements Re-transmissions In-order delivery DoS attacks Byte-oriented

70 SCTP Stream Control Transmission Protocol: RFC 4960 Transport protocol that provides similar service as TCP, but overcoming some of TCPs limitations Reliable data transfer and strict order-of-transmission delivery of data can cause unnecessary delay Stream-oriented nature of TCP is inconvenient TCP sockets do not enable multi-homed hosts TCP is vulnerable to SYN attacks

71 SCTP Functional Overview

72 SCTP Features Message oriented (not byte) Chunks, identified by header inside each packet Provides reliable transmission Re-transmission, re-ordering Supports multiple streams in one connection Supports multi-homing (one connection, multiple IP) Resilience against SYN flood attacks 4-way handshake

73 SCTP 4-way Handshake SYN flood attacks are a major problem with TCP Servers allocate resources to a connection before the initiation handshake is finished

74 SCTP 4-way Handshake

75 SCTP 4-way Handshake SCTP only allocates resources to a connection after he receives back a cookie sent out in the answer to the INIT request Receiver does not need to decrypt the cookie, so no key exchange is needed To reduce the overhead, data can be sent with the Cookie Echo and Cookie ACK chunks

76 Datagram Congestion Control Protocol Avoid congestion collapse by providing an easyto-use congestion control for long-lived flows without reliability While Having no start-up delay Being stateless Being timely instead of reliable Leaving some control over the congestion algorithm to the application

77 Why a new transport protocol? Change TCP? Too many unforeseen interactions High overhead Unreliable variants of SCTP? High overhead Only TCP-like congestion control Little flexibility

78 Why a new transport protocol? Change RTP? RTP is mostly implemented in the applications Not all RTP functionality is needed RTP would need to access information in the UDP and IP headers It is better to design a light-weight protocol from scratch that offers only congestion control

79 Protocol Requirements Protocol must provide set of congestion control mechanisms that application can choose from Congestion control mechanism must be reliably negotiated between endpoints Support for ECN should be included Low overhead DoS protection for servers Connection setup and teardown to pass firewalls

80 DCCP Handshake REQ Response ACK DATA ACK Close Req Close Reset

81 DCCP Header Res Data Offset Type Source Port X Reserved CCVal Reserved CsCov Sequence Number (low) ACK Sequence Number (low) Destination Port Checksum Sequence Number ACK Sequence Number Optional

82 DCCP Sequence Numbers Sequence number identify packets (not bytes) Every packet uses sequence number space All connection management packets must set X=1 and use the 48-bit sequence numbers For security reasons Each new packet has the sequence number of the previous packet plus 1 If X=0, 16-bit Sequence Number is used and header length is 12 Bytes only

83 DCCP Overhead? 48-bit sequence numbers => 18 bytes header If X=0, only upper 16 bits of Sequence Number are used and header length is 12 Bytes only Calculate resulting overhead for a G.711 VoIP packet!

84 DCCP Acknowledgements Header acknowledge carries sequence number of last received packet TCP acknowledges earliest not received packet Detailed information about previous packets can be sent as option AckVector reports which packets were received and whether they were ECN-marked

85 DCCP: RFCs RFC 4336: Problem statement for the datagram congestion control protocol, S. Floyd, M. Handley, E. Kohler RFC 4340: Datagram Congestion Control Protocol, E. Kohler, M. Handley, S. Floyd

86 Recommended Reading SCTP: State-of-the-art in research, products and technical challenges, S. Fu, M. Atiquzzaman, IEEE Communications Magazine, 2004 Designing DCCP: congestion control without reliability, E. Kohler, M. Handley, S. Floyd, Sigcomm 06, 2006 Low latency adaptive streaming over TCP, A. Goel, C. Krasic, J. Walpole, Transactions on Multimedia Computing, Communications and Applications, Aug 2008

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