Transcript
Multimedia Networking
Adapted from slides of: J.F Kurose and K.W. Ross (copyright 1996-2006), Shivkumar Kalyanaraman, Henning Schulzrinne, Doug Moeller, Francesco Santini
Multimedia, Quality of Service: What is it? Multimedia applications: network audio and video (“continuous media”)
QoS network provides application with level of
performance needed for application to function.
Goals Principles Classify multimedia applications Identify the network services the apps need Making the best of best effort service Mechanisms for providing QoS Protocols and Architectures Specific protocols for best-effort Architectures for QoS VoIP services Specific architectures and protocols Implementation issues
MM Networking Applications Classes of MM applications: 1) Streaming stored audio and video 2) Streaming live audio and video 3) Real-time interactive audio and video
Jitter is the variability of packet delays within the same packet stream
Fundamental characteristics: Typically delay sensitive
end-to-end delay delay jitter
But loss tolerant:
infrequent losses cause minor glitches Antithesis of data, which are loss intolerant but delay tolerant.
Streaming Stored Multimedia
Streaming: media stored at source transmitted to client streaming: client playout begins before all data has arrived timing constraint for still-to-be
transmitted data: in time for playout
Streaming Stored Multimedia: What is it?
1. video recorded
2. video sent
network delay
3. video received, played out at client
streaming: at this time, client
playing out early part of video, while server still sending later part of video
time
Streaming Stored Multimedia: Interactivity
VCR-like functionality: client can
pause, rewind, FF, push slider bar 10 sec initial delay OK 1-2 sec until command effect OK RTSP often used (more later) timing constraint for still-to-be transmitted data: in time for playout
Streaming Live Multimedia Examples: Internet radio talk show Live sporting event Streaming playback buffer playback can lag tens of seconds after transmission still have timing constraint Interactivity fast forward impossible rewind, pause possible!
Interactive, Real-Time Multimedia
applications: IP telephony,
video conference, distributed interactive worlds end-end delay requirements: audio: < 150 msec good, < 400 msec OK • includes application-level (packetization) and network delays • higher delays noticeable, impair interactivity
session initialization
how does callee advertise its IP address, port number, encoding algorithms?
Multimedia Over Today’s Internet TCP/UDP/IP: “best-effort service”
no guarantees on delay, loss
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But you said multimedia apps requires ? QoS and level of performance to be ? ? effective! ?
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Today’s Internet multimedia applications use application-level techniques to mitigate (as best possible) effects of delay, loss
How should the Internet evolve to better support multimedia? Integrated services philosophy: Fundamental changes in Internet so that apps can reserve end-to-end bandwidth Requires new, complex software in hosts & routers Laissez-faire no major changes more bandwidth when needed content distribution, application-layer multicast
application layer
Differentiated services philosophy: Fewer changes to Internet infrastructure, yet provide 1st and 2nd class service.
What’s your opinion?
A few words about audio compression Analog signal sampled
at constant rate
telephone: 8,000 samples/sec CD music: 44,100 samples/sec
Each sample quantized,
i.e., rounded
e.g., 28=256 possible quantized values
Each quantized value
represented by bits
8 bits for 256 values
Example: 8,000
samples/sec, 256 quantized values --> 64,000 bps Receiver converts it back to analog signal:
some quality reduction
Example rates CD: 1.411 Mbps MP3: 96, 128, 160 kbps Internet telephony: 5.3 - 13 kbps
A few words about video compression Video is sequence of
images displayed at constant rate
e.g. 24 images/sec
Digital image is array of
pixels Each pixel represented by bits Redundancy
spatial temporal
Examples: MPEG 1 (CD-ROM) 1.5 Mbps MPEG2 (DVD) 3-6 Mbps MPEG4 (often used in Internet, < 1 Mbps) Research: Layered (scalable) video
adapt layers to available bandwidth
Streaming Stored Multimedia Application-level streaming techniques for making the best out of best effort service: client side buffering use of UDP versus TCP multiple encodings of multimedia
Media Player jitter removal decompression error concealment
graphical user interface
w/ controls for interactivity
Internet multimedia: simplest approach
audio or video stored in file
files transferred as HTTP object
received in entirety at client then passed to player
audio, video not streamed: no, “pipelining,” long delays until playout!
Internet multimedia: streaming approach
browser GETs metafile browser launches player, passing metafile player contacts server server streams audio/video to player
Streaming from a streaming server
This architecture allows for non-HTTP protocol between
server and media player Can also use UDP instead of TCP.
Streaming Multimedia: Client Buffering
variable network delay
client video reception
constant bit rate video playout at client
buffered video
constant bit rate video transmission
client playout delay
Client-side buffering, playout delay compensate
for network-added delay, delay jitter
time
Streaming Multimedia: Client Buffering
constant drain rate, d
variable fill rate, x(t)
buffered video
Client-side buffering, playout delay compensate
for network-added delay, delay jitter
Streaming Multimedia: UDP or TCP? UDP server sends at rate appropriate for client (oblivious to
network congestion !) often send rate = encoding rate = constant rate then, fill rate = constant rate - packet loss short playout delay (2-5 seconds) to compensate for network delay jitter error recover: time permitting
TCP send at maximum possible rate under TCP fill rate fluctuates due to TCP congestion control larger playout delay: smooth TCP delivery rate HTTP/TCP passes more easily through firewalls
Streaming Multimedia: client rate(s) 1.5 Mbps encoding
28.8 Kbps encoding
Q: how to handle different client receive rate capabilities? 28.8 Kbps dialup 100Mbps Ethernet A: server stores, transmits multiple copies of video, encoded at different rates
User Control of Streaming Media: RTSP HTTP Does not target multimedia content No commands for fast forward, etc. RTSP: RFC 2326 Client-server application layer protocol. For user to control display: rewind, fast forward, pause, resume, repositioning, etc…
What it doesn’t do: does not define how audio/video is encapsulated for streaming over network does not restrict how streamed media is transported; it can be transported over UDP or TCP does not specify how the media player buffers audio/video
RTSP: out of band control FTP uses an “out-of-band” control channel: A file is transferred over one TCP connection. Control information (directory changes, file deletion, file renaming, etc.) is sent over a separate TCP connection. The “out-of-band” and “inband” channels use different port numbers.
RTSP messages are also sent out-of-band: RTSP control messages use different port numbers than the media stream: out-of-band.
Port 554
The media stream is
considered “in-band”.
RTSP Example Scenario: metafile communicated to web browser
browser launches player player sets up an RTSP control connection, data
connection to streaming server
Metafile Example
Twister
RTSP Operation
RTSP Exchange Example C: SETUP rtsp://audio.example.com/twister/audio RTSP/1.0 Cseq=1; Transport: rtp/udp; compression; port=3056; mode=PLAY S: RTSP/1.0 200 OK Cseq=1 Session 4231 C: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Cseq=2; Session: 4231 Range: npt=0S: RTSP/1.0 200 OK Cseq=2 Session 4231 C: PAUSE rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Cseq=3; Session: 4231 Range: npt=37 S: RTSP/1.0 200 OK Cseq=3; Session 4231 C: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Cseq=4; Session: 4231 S: RTSP/1.0 200 OK Cseq=4; Session 4231
Real-time interactive applications PC-2-PC phone instant messaging services are providing this PC-2-phone
Dialpad Net2phone videoconference with Webcams
Going to now look at a PC-2-PC Internet phone example in detail
Interactive Multimedia: Internet Phone Introduce Internet Phone by way of an example speaker’s audio: alternating talk spurts, silent
periods.
64 kbps during talk spurt
pkts generated only during talk spurts
20 msec chunks at 8 Kbytes/sec: 160 bytes data
application-layer header added to each chunk. Chunk+header encapsulated into UDP segment. application sends UDP segment into socket every
20 msec during talkspurt.
Internet Phone: Packet Loss and Delay network loss: IP datagram lost due to network
congestion (router buffer overflow) delay loss: IP datagram arrives too late for playout at receiver
delays: processing, queueing in network; end-system (sender, receiver) delays typical maximum tolerable delay: 400 ms
loss tolerance: depending on voice encoding, losses
concealed, packet loss rates between 1% and 10% can be tolerated.
Delay Jitter
variable network delay (jitter)
client reception
constant bit rate playout at client
buffered data
constant bit rate transmission
client playout delay
time
Consider the end-to-end delays of two consecutive
packets: difference can be more or less than 20 msec
Internet Phone: Fixed Playout Delay Receiver attempts to playout each chunk exactly q
msecs after chunk was generated. chunk has time stamp t: play out chunk at t+q . chunk arrives after t+q: data arrives too late for playout, data “lost” Tradeoff for q: large q: less packet loss small q: better interactive experience
Fixed Playout Delay • Sender generates packets every 20 msec during talk spurt.
• First packet received at time r • First playout schedule: begins at p • Second playout schedule: begins at p’ packets
loss
packets generated packets received
playout schedule p' - r playout schedule p-r
time r p
p'
Adaptive Playout Delay, I Goal: minimize playout delay, keeping late loss rate low Approach: adaptive playout delay adjustment:
Estimate network delay, adjust playout delay at beginning of each talk spurt. Silent periods compressed and elongated. Chunks still played out every 20 msec during talk spurt.
t i timestamp of the ith packet ri the time packet i is received by receiver p i the time packet i is played at receiver ri t i network delay for ith packet d i estimate of average network delay after receiving ith packet
Dynamic estimate of average delay at receiver:
di (1 u)di 1 u(ri ti ) where u is a fixed constant (e.g., u = .01).
Adaptive playout delay II Also useful to estimate the average deviation of the delay, vi :
vi (1 u)vi 1 u | ri ti di | The estimates di and vi are calculated for every received packet, although they are only used at the beginning of a talk spurt. For first packet in talk spurt, playout time is:
pi ti di Kvi where K is a positive constant. Remaining packets in talkspurt are played out periodically
Adaptive Playout, III Q: How does receiver determine whether packet is first in a talkspurt? If no loss, receiver looks at successive timestamps.
difference of successive stamps > 20 msec -->talk spurt begins.
With loss possible, receiver must look at both time
stamps and sequence numbers.
difference of successive stamps > 20 msec and sequence numbers without gaps --> talk spurt begins.
Recovery from packet loss (1) forward error correction (FEC): simple scheme for every group of n chunks create a redundant chunk by exclusive OR-ing the n original chunks send out n+1 chunks, increasing the bandwidth by factor 1/n. can reconstruct the original n chunks if there is at most one lost chunk from the n+1 chunks
Playout delay needs to
be fixed to the time to receive all n+1 packets Tradeoff: increase n, less bandwidth waste increase n, longer playout delay increase n, higher probability that 2 or more chunks will be lost
Recovery from packet loss (2) 2nd FEC scheme • “piggyback lower quality stream” • send lower resolution audio stream as the redundant information • for example, nominal stream PCM at 64 kbps and redundant stream GSM at 13 kbps. • Whenever there is non-consecutive loss, the
receiver can conceal the loss. • Can also append (n-1)st and (n-2)nd low-bit rate chunk
Recovery from packet loss (3)
Interleaving chunks are broken up into smaller units for example, 4 5 msec units per chunk Packet contains small units from different chunks
if packet is lost, still have
most of every chunk has no redundancy overhead but adds to playout delay
Real-Time Protocol (RTP) RTP specifies a packet
structure for packets carrying audio and video data RFC 1889. RTP packet provides
payload type identification packet sequence numbering timestamping
RTP runs in the end
systems. RTP packets are encapsulated in UDP segments Interoperability: If two Internet phone applications run RTP, then they may be able to work together
RTP runs on top of UDP RTP libraries provide a transport-layer interface that extend UDP: • port numbers, IP addresses • payload type identification • packet sequence numbering • time-stamping
RTP Example Consider sending 64
kbps PCM-encoded voice over RTP. Application collects the encoded data in chunks, e.g., every 20 msec = 160 bytes in a chunk. The audio chunk along with the RTP header form the RTP packet, which is encapsulated into a UDP segment.
RTP header indicates
type of audio encoding in each packet
sender can change encoding during a conference.
RTP header also
contains sequence numbers and timestamps.
RTP and QoS RTP does not provide any mechanism to ensure
timely delivery of data or provide other quality of service guarantees. RTP encapsulation is only seen at the end systems: it is not seen by intermediate routers.
Routers providing best-effort service do not make any special effort to ensure that RTP packets arrive at the destination in a timely matter.
RTP header 0
V
2
P
3 4 X
CC
8 9 M
PT
16
Timestamp SSRC CSRC_1
24
Sequence Number
31
RTP header version (V) = (2 bit)
RTP protocol version;
Padding (P) = (1 bit)
Indicates the presence of padding bytes beyond data bytes;
Extension (X) = (1 bit)
Indicates the presence of an extension header;
CSRC count (CC) = (4 bit)
Number of CSRC fields in the header, i.e. number of sources that have generated data included in paylaod;
Marker (M) = (1 bit)
May be used by applications, for example to indicate end of data.
RTP header Payload Type (7 bits): Indicates type of encoding currently being used. If sender changes encoding in middle of conference, sender informs the receiver through this payload type field. •Payload type 0: PCM mu-law, 64 kbps •Payload type 3, GSM, 13 kbps •Payload type 7, LPC, 2.4 kbps •Payload type 26, Motion JPEG •Payload type 31. H.261 •Payload type 33, MPEG2 video
Sequence Number (16 bits): Increments by one for each RTP packet sent, and may be used to detect packet loss and to restore packet sequence. At the session start its value is set randomly, so that the probability of mixing packets of different sessions is minimized.
RTP header Timestamp field (32 bytes long). Reflects the sampling instant of the first byte in the RTP data packet. For audio, timestamp clock typically increments by one for each sampling period (for example, each 125 usecs for a 8 KHz sampling clock) if application generates chunks of 160 encoded samples, then timestamp increases by 160 for each RTP packet when source is active. Timestamp clock continues to increase at constant rate when source is inactive.
SSRC field (32 bits long). Identifies the source of the RTP stream. In case of multiple source, the SSRC source has mixed data. Contributing source (CSRC) = up to 15 fields, 32 bits each. - optional fields; - they include the SSRC of the true flow sources.
RTP header 32 bit Source port #
Destination port #
Lenght
Checksum (opt.)
V P X
C C
M
PType
Sequence number
Timestamp Synchronization source (SSRC) identifier Possible header extension
Payload
UDP header 8B
RTP header 12 B
RTP header 0
8
Defined by profile
16
24
Lenght
31
header extension ……..
header extension used for individual implementations to test new
features, payload formats, requiring information that cannot be included in the normal RTP header. lenght : header extension length expressed in 4 bytes words.
Real-Time Control Protocol (RTCP) Works in conjunction with
RTP. Each participant in RTP session periodically transmits RTCP control packets to all other participants. Each RTCP packet contains sender and/or receiver reports
report statistics useful to application
Statistics include number
of packets sent, number of packets lost, interarrival jitter, etc. Feedback can be used to control performance Sender may modify its transmissions based on feedback
RTCP - Continued
- For an RTP session there is typically a single multicast address; all RTP and RTCP packets belonging to the session use the multicast address. - RTP and RTCP packets are distinguished from each other through the use of distinct port numbers.
- To limit traffic, each participant reduces his RTCP traffic as the number of conference participants increases.
RTCP Packets Receiver report packets Sender report packets
Source description packets: e-mail address of sender, sender's name, SSRC of associated RTP stream. Provide mapping between the SSRC and the user/host name.
RTCP Packets
SR (Sender Report):
- sent from all active sources to all participants, i.e. SSRC of the RTP stream; - it includes transmission statistics collected by SSRCs - it includes information related to data sent: a) timestamp (NTP) of the sending time;
b) timestamp relevant to the ongoing RTP flow; c) data sent since the session start: - total number of RTP packets sent; - total number of bytes sent.
RTCP Packets RR (Receiver Report):
- sent from all passive terminals to all participants; - it includes reception statistics collected by a partecipanti receiving RTP data; - used to inform senders of the reception quality; - sent to all sources form which a SR has been received; - it includes: a) Indication of the transmistting suorce; b) Timestamp of the latter SR received; c) Reception delay of the latter SR; d) Highest sequence number received from the source; e) Number of lost RTP packets in the session; f) Fraction of lost RTP packets in the session; g) jitter estimate of RTP packets in the session.
RTCP Packets SDES (Source Descriptor):
- description of RTP participants, - provide mapping between the SSRC and the user/host name; - unique identifier; - used by sources and receivers to present themselves. - it may include: a) CNAME: user id(
[email protected]); b) NAME: name of the person using the appliation; c) EMAIL; d) PHONE; e) LOC: user geographical location; f) TOOL: application using RTP; g) NOTE.
BYE:
- it can indicate a participant disconnection or the session end.
APP: application-specific
- it indicates that a participant wants to leave the session.
RTCP Packets Encription prefix 32 bit
SR o RR
Additional RRs
SDES (CNAME)
APP
BYE
Synchronization of Streams RTCP can synchronize
different media streams within a RTP session. Consider videoconferencing app for which each sender generates one RTP stream for video and one for audio. Timestamps in RTP packets tied to the video and audio sampling clocks not tied to the wallclock time
Each RTCP sender-report
packet contains (for the most recently generated packet in the associated RTP stream):
timestamp of the RTP packet wall-clock time for when packet was created.
Receivers can use this
association to synchronize the playout of audio and video.
RTCP Bandwidth Scaling Problem !!! Consider an RTP session, with one sender and many receivers; Each receiver generates RTCP packets; the aggregate receiver data transmission rate may be higher than the sender data transmission rate. The amount of RTP traffic sent through the multicast tree does not
change with the receiver number;
The amount of RTCP traffic increases linearly with the receiver number.
Solution: RTCP adapts the transmission rate of the session participants.
RTCP Bandwidth Scaling RTCP attempts to limit its
The 75 kbps is equally shared
traffic to 5% of the among receivers: session bandwidth. With R receivers, each Example receiver gets to send RTCP traffic at 75/R kbps. Suppose one sender, sending video at a rate of 2 Sender gets to send RTCP Mbps. Then RTCP attempts traffic at 25 kbps. to limit its traffic to 100 Participant determines RTCP Kbps. packet transmission period by RTCP gives 75% of this calculating avg RTCP packet rate to the receivers; size (across the entire remaining 25% to the session) and dividing by sender allocated rate.
RTCP Bandwidth Scaling Each participant (sender or receiver) determines the
transmission time of an RTCP packet by evaluation dinamically the average RTCP packet size and dividing it by the allocated transmissio rate.
Ts =
number of senders
(ave. RTCP packet size)
0.25 x 0.05 x session bandwidth
Tr =
number of receivers 0.75 x 0.05 x session bandwidth
(ave. RTCP packet size)
SIP
SIP Session Initiation Protocol Comes from IETF
SIP long-term vision All telephone calls and video conference calls take place over the Internet People are identified by names or e-mail addresses, rather than by phone numbers. You can reach the callee, no matter where the callee roams, no matter what IP device the callee is currently using.
SIP Services Setting up a call Provides mechanisms for caller to let callee know she wants to establish a call Provides mechanisms so that caller and callee can agree on media type and encoding. Provides mechanisms to end call.
Determine current IP
address of callee.
Maps mnemonic identifier to current IP address
Call management Add new media streams during call Change encoding during call Invite others Transfer and hold calls
H.323 vs SIP SIP: IETF standard
Derived from HTTP style signaling, Simple and interfaces well with IP networks, instant messaging (IM) Services are not explicitly exposed to protocol Well-defined methods can be used to design services: most telephony services have analogs in the SIP world today SIP is gathering market share rapidly
SIP Audio Codec
Video Codec
G.711
H.261
G.723
H.263
G.729
RTP
SIP TCP
UDP
IP
LAN Interface
RTCP
SIP functionality IETF-standardized
peer-to-peer signaling protocol (RFC
2543): Locate user given email-style address Setup session (call) (Re)-negotiate call parameters Manual and automatic forwarding Personal mobility: different terminal, same identifier Terminate and transfer calls
IP SIP Phones and Adaptors 1
Are true Internet hosts • Choice of application
• Choice of server
Analog phone adaptor
2
• IP appliances Implementations • 3Com (3)
3
• Columbia University Palm control
• MIC WorldCom (1) • Mediatrix (1) • Nortel (4) • Siemens (5)
44
5
SIP components UAC: user-agent client (caller application) UAS: user-agent server: accept, redirect, refuse
call redirect server: redirect requests proxy server: server + client registrar: track user locations user agent = UAC + UAS often combine registrar + (proxy or redirect server)
Setting up a call to a known IP address Bob
Alice
167.180.112.24 INVITE bob @193.64.2 10.89 c=IN IP4 16 7.180.112.2 4 m=audio 38 060 RTP/A VP 0
193.64.210.89
port 5060
port 5060
Bob's terminal rings
200 OK .210.89 c=IN IP4 193.64 RTP/AVP 3 3 75 m=audio 48
ACK
port 5060
• Alice’s SIP invite message indicates her port number & IP address. Indicates encoding that Alice prefers to receive (PCM ulaw) • Bob’s 200 OK message indicates his port number, IP address & preferred encoding (GSM)
m Law audio port 38060
GSM
time
port 48753
time
• SIP messages can be sent over TCP or UDP; here sent over RTP/UDP. •Default SIP port number is 5060.
Setting up a call (more) Codec negotiation:
Suppose Bob doesn’t have PCM ulaw encoder. Bob will instead reply with 606 Not Acceptable Reply and list encoders he can use. Alice can then send a new INVITE message, advertising an appropriate encoder.
Rejecting the call
Bob can reject with replies “busy,” “gone,” “payment required,” “forbidden”. Media can be sent over RTP or some other protocol.
Example of SIP message INVITE sip:
[email protected] SIP/2.0 Via: SIP/2.0/UDP 167.180.112.24 From: sip:
[email protected] To: sip:
[email protected] Call-ID:
[email protected] Content-Type: application/sdp Content-Length: 885 c=IN IP4 167.180.112.24 m=audio 38060 RTP/AVP 0 Notes: HTTP message syntax sdp = session description protocol Call-ID is unique for every call.
• Here we don’t know
Bob’s IP address. Intermediate SIP servers will be necessary. • Alice sends and
receives SIP messages using the SIP default port number 506. • Alice specifies in Via: header that SIP client sends and receives SIP messages over UDP
Name translation and user locataion Caller wants to call
callee, but only has callee’s name or e-mail address. Need to get IP address of callee’s current host:
user moves around DHCP protocol user has different IP devices (PC, PDA, car device)
Result can be based on: time of day (work, home) caller (don’t want boss to call you at home) status of callee (calls sent to voicemail when callee is already talking to someone)
Service provided by SIP servers: SIP registrar server SIP proxy server
SIP Registrar When Bob starts SIP client, client sends SIP
REGISTER message to Bob’s registrar server (similar function needed by Instant Messaging)
Register Message: REGISTER sip:domain.com SIP/2.0 Via: SIP/2.0/UDP 193.64.210.89 From: sip:
[email protected] To: sip:
[email protected] Expires: 3600
SIP: Personal Mobility Users maintain a single externally visible identifier regardless of their network location
SIP Proxy Alice sends invite message to her proxy server contains address sip:
[email protected] Proxy responsible for routing SIP messages to
callee
possibly through multiple proxies.
Callee sends response back through the same set
of proxies. Proxy returns SIP response message to Alice
contains Bob’s IP address
Note: proxy is analogous to local DNS server
Example Caller
[email protected] with places a call to
[email protected]
SIP registrar upenn.edu SIP registrar eurecom.fr
2
(1) Jim sends INVITE message to umass SIP proxy. (2) Proxy forwards request to upenn registrar server. (3) upenn server returns redirect response, indicating that it should try
[email protected]
SIP proxy umass.edu
1
3
4
5
7 8
6
9
SIP client 217.123.56.89
SIP client 197.87.54.21
(4) umass proxy sends INVITE to eurecom registrar. (5) eurecom registrar forwards INVITE to 197.87.54.21, which is running keith’s SIP client. (6-8) SIP response sent back (9) media sent directly between clients. Note: also a SIP ack message, which is not shown.
SIP as Event Notification Protocol
SIP for instant messaging: IM (RFC 3428) IM: transfer of (short) messages in near real-time, for
conversational mode. Current IM: proprietary, server-based and linked to buddy lists etc MESSAGE method: inherits SIP’s request routing and security features Message content as MIME body parts Sent in the context of some SIP dialog (note: slightly different from pager mode: asynchronous) Sent over TCP (or congestion controlled transports): lots of messaging volumes… Allows IM applications to potentially interoperate and also provides SIP-based integration with other multimedia streams.
SIP: Presence
SIP-based Architecture rtspd
RTSP media server
RTSP
sipconf
Telephone
SIP conference server
Telephone switch
T1/E1 RTP/SIP
sipd SIP proxy, redirect server
RTSP clients sipum SIP/RTSP Unified messaging
SQL database
Web based configuration
Web server e*phone
Cisco 2600 gateway
Hardware Internet (SIP) phones
sipc
NetMeeting
sip323 Software SIP user agents
Quicktime
SIPH.323 convertor
H.323
Example Call • sipd canonicalizes the destination to
• Bob signs up for the service from the web as “
[email protected]”
sip:
[email protected]
• sipd rings both e*phone and sipc
• He registers from multiple phones • Alice tries to reach Bob INVITE ip:
[email protected]
• Bob accepts the call from sipc and starts talking Web based configuration
sipd SIP proxy, redirect server
Call Bob
SQL database
e*phone Hardware Internet (SIP) phones
sipc
ecse.rpi.edu Software SIP user agents
Web server
Proxy Server 1. INVITE sip:
[email protected] SIP/2.0 From: sip:
[email protected] 2. INVITE sip:dcheney@wh SIP/2.0 From: sip:
[email protected] 3. SIP/2.0 200 ok From: sip:dcheney@wh
parliament.uk
us.gov
Location Server
[email protected]
4
george.w.bush
1&5
2&6 4. SIP/2.0 100 OK From: sip:
[email protected] 5. ACK sip:
[email protected] SIP/2.0 From: sip:
[email protected] 6. ACK sip:dcheney@wh SIP/2.0 From: sip:
[email protected]
Proxy server
dcheney@wh
3
Redirect Server us.gov
parliament.uk
george.w.bush
Location Server
1&3
2
[email protected]
[email protected] Redirect Server
4&6 5 1. INVITE sip:
[email protected] From: sip:
[email protected] 2. SIP/2.0 320 Moved temporarily Contact: sip:
[email protected] 3. ACK sip:
[email protected] From: sip:
[email protected]
4. INVITE sip:
[email protected]. ACK sip:
[email protected] From:
[email protected] From: sip:
[email protected] 5. SIP/2.0 200 OK To:
[email protected]
SIP Call Signaling Assumes Endpoints(Clients) know each other’s IP addresses SIP Endpoint Signaling Plane
SIP Gateway Invite 180 Ringing 200 OK
SIP + SDP (TCP or UDP)
Ack
Bearer Plane
RTP Stream RTP Stream RTCP Stream
Media (UDP)
SDP: Session Description Protocol – RFC 2327
Not really a
protocol – describes data carried by
other protocols Used by SAP, SIP, RTSP, H.332, PINT.
Session description v= (protocol version) o= (owner/creator and session identifier). s= (session name) i=* (session information) u=* (URI of description) e=* (email address) p=* (phone number) c=* (connection information - not required if included in all media) b=* (bandwidth information) One or more time descriptions z=* (time zone adjustments) k=* (encryption key) a=* (zero or more session attribute lines) Zero or more media descriptions)
Time description t= (time the session is active) r=* (zero or more repeat times) Media description m= (media name and transport address) i=* (media title) c=* (connection information - optional if included at session-level) b=* (bandwidth information) k=* (encryption key) a=* (zero or more media attribute lines)
SDP: Session Description Protocol – RFC 2327 Example:
o=
< network type > version number f or this announcement
internet
v=0 o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4 s=SDP Seminar i=A Seminar on the session description protocol u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps [email protected] (Mark Handley) URI should be a pointer to additional information about the conference c=IN IP4 224.2.17.12/127 t=2873397496 2873404696 Start and End time in NTP a=recvonly m=audio 49170 RTP/AVP 0format m=video 51372 RTP/AVP 31 m=application 32416 udp wb a=orient:portrait
SIP Dialogs (RFC 3261) A dialog represents a peer-to-peer SIP relationship between
two user agents that persists for some time. The dialog facilitates sequencing of messages between the user agents and proper routing of requests between both of them. The dialog represents a context in which to interpret SIP messages. A dialog is identified at each UA with a dialog ID, which consists of a Call-ID value, a local tag and a remote tag. A dialog contains certain pieces of state needed for further message transmissions within the dialog.
UPDATE method (RFC 3311) INVITE method: initiation and modification of sessions.
INVITE affects two pieces of state: session (the media streams SIP sets up) and dialog (the state that SIP itself defines). Issue: need to modify session aspects before the initial INVITE has been answered. A re-INVITE cannot be used for this purpose: impacts the state of the dialog, in addition to the session. Ans: The UPDATE method Operation: (Offer/Answer model) The caller begins with an INVITE transaction, which proceeds normally. Once a dialog is established, either early or confirmed, … … the caller can generate an UPDATE method that contains an SDP offer for the purposes of updating the session. The response to the UPDATE method contains the answer. Similarly, once a dialog is established, the callee can send an UPDATE offer
Content distribution networks (CDNs) Content replication Challenging to stream large
files (e.g., video) from single origin server in real time Solution: replicate content at hundreds of servers throughout Internet content downloaded to CDN servers ahead of time placing content “close” to user avoids impairments (loss, delay) of sending content over long paths CDN server typically in edge/access network
origin server in North America
CDN distribution node
CDN server in S. America CDN server in Europe
CDN server in Asia
CDN example
HTTP request for www.foo.com/sports/sports.html
Origin server
1 2 3
DNS query for www.cdn.com
CDNs authoritative DNS server HTTP request for www.cdn.com/www.foo.com/sports/ruth.gif
Nearby CDN server
origin server (www.foo.com) distributes HTML replaces: http://www.foo.com/sports.ruth.gif
with http://www.cdn.com/www.foo.com/sports/ruth.gif
CDN company (cdn.com) distributes gif files uses its authoritative DNS server to route redirect requests
More about CDNs routing requests CDN creates a “map”, indicating distances from leaf ISPs and CDN nodes when query arrives at authoritative DNS server: server determines ISP from which query originates
uses “map” to determine best CDN server
CDN nodes create application-layer overlay
network
Improving QOS in IP Networks Thus far: “making the best of best effort” Future: next generation Internet with QoS guarantees RSVP: signaling for resource reservations Differentiated Services: differential guarantees Integrated Services: firm guarantees simple model for sharing and congestion studies:
Principles for QOS Guarantees Example: 1MbpsI P phone, FTP share 1.5 Mbps link. bursts of FTP can congest router, cause audio loss want to give priority to audio over FTP
Principle 1 packet marking needed for router to distinguish between different classes; and new router policy to treat packets accordingly
Principles for QOS Guarantees (more) what if applications misbehave (audio sends higher
than declared rate)
policing: force source adherence to bandwidth allocations
marking and policing at network edge:
Principle 2 provide protection (isolation) for one class from others
Principles for QOS Guarantees (more) fixed (non-sharable) bandwidth to flow: inefficient use of bandwidth if flows doesn’t use
Allocating
its allocation
Principle 3 While providing isolation, it is desirable to use resources as efficiently as possible
Principles for QOS Guarantees (more)
Basic fact of life: can not support traffic demands beyond link capacity
Principle 4 Call Admission: flow declares its needs, network may block call (e.g., busy signal) if it cannot meet needs
Summary of QoS Principles
Let’s next look at mechanisms for achieving this ….
Scheduling And Policing Mechanisms scheduling: choose next packet to send on link FIFO (first in first out) scheduling: send in order of
arrival to queue
real-world example? discard policy: if packet arrives to full queue: who to discard? • Tail drop: drop arriving packet • priority: drop/remove on priority basis • random: drop/remove randomly
Scheduling Policies: more Priority scheduling: transmit highest priority queued packet multiple classes, with different priorities
class may depend on marking or other header info, e.g. IP source/dest, port numbers, etc.. Real world example?
Scheduling Policies: still more round robin scheduling: multiple classes cyclically scan class queues, serving one from each class (if available) real world example?
Scheduling Policies: still more Weighted Fair Queuing: generalized Round Robin each class gets weighted amount of service in each cycle real-world example?
Policing Mechanisms Goal: limit traffic to not exceed declared parameters Three common-used criteria:
(Long term) Average Rate: how many pkts can be sent per unit time (in the long run)
crucial question: what is the interval length: 100 packets per sec or 6000 packets per min have same average!
Peak Rate: to be defined according a reference time slot.
(Max.) Burst Size: max. number of pkts sent consecutively (at peak rate)
Policing Mechanisms Token Bucket: limit input to specified Burst Size and Average Rate.
bucket can hold b tokens tokens generated at rate
full
r token/sec unless bucket
over interval of length t: number of packets admitted less than or equal to (r t + b).
Provisioning and Monitoring LB
BTS
LB
Ps
rs
BTS
rs
Ps
Dual Leaky-Bucket b c
BTS
rs
Ps c BTS b Ps rs
Ps
b
BTS
rs
Ps
c
Multiplexing DLB1
B C
DLB2 ...
Ex. Fair-Share b
DLBk
C B c b
c
Equivalent Bandwidth and Buffer b Ps c BTS b Ps rs
BTS
Fair-Share
PS BTS c0 Dmax ( PS rS ) BTS B Dmax C C B nmax c0 b0
or
b0
C B c b
b0 Dmax c0
C B c b
b Dmax c
Max delay rs
c0
Ps
c
b Dmax c PS BTS c0 Dmax ( PS rS ) BTS b0 Dmax c0 B C nmax min , b0 c0 B typically Dmax C
Policing Mechanisms (more) token bucket, WFQ combine to provide guaranteed
upper bound on delay, i.e., QoS guarantee!
arriving traffic
token rate, r bucket size, b
WFQ
per-flow rate, R
D = b/R max
IETF Integrated Services architecture for providing QOS guarantees in IP
networks for individual application sessions resource reservation: routers maintain state info (a la VC) of allocated resources, QoS req’s admit/deny new call setup requests: Question: can newly arriving flow be admitted with performance guarantees while not violated QoS guarantees made to already admitted flows?
Intserv: QoS guarantee scenario Resource reservation call setup, signaling (RSVP) traffic, QoS declaration per-element admission control
request/ reply
QoS-sensitive scheduling (e.g., WFQ)
Integrated Services - Principles Flow specification
Tell the network what the flow wants e.g. 100 msec guaranteed to www.nsf.gov Admission control Network decides if it can handle flow Spec travels down path for approval Delay guarantee approved by all routers, so admitted Reservation Enable admission control Packet classification Map packets to flows e.g. packets marked as guaranteed Scheduling • Forwarding policy • e.g. guaranteed packets sent first
Intserv QoS: Service models [RFC 2211, RFC 2212] Controlled load service:
Guaranteed service:
"a quality of service closely
worst case traffic arrival:
approximating the QoS that same flow would receive from an unloaded network element."
leaky-bucket-policed source simple (mathematically provable) bound on delay [Parekh 1992, Cruz 1988] arriving traffic
token rate, r bucket size, b
WFQ
per-flow rate, R
D = b/R max
Call Admission Arriving session must : declare its QOS requirement R-spec:
defines the QOS being requested characterize traffic it will send into network T-spec: defines traffic characteristics signaling protocol: needed to carry R-spec and Tspec to routers (where reservation is required) RSVP
Advanced IP Architecture Real Time Applications
RSVP
RTP/RTCP
Elastic Applications
UDP IPv4/IPv6
Underlying Data Link Technologies
TCP
Role of RSVP Rides on top of unicast/multicast routing
protocols Must be present at sender(s), receiver(s), and routers Carries resource requests all the way through the network At each hop consults admission control and sets up reservation
RSVP Design Goals 1. 2. 3. 4.
5. 6.
accommodate heterogeneous receivers (different bandwidth along paths) accommodate different applications with different resource requirements make multicast a first class service, with adaptation to multicast group membership leverage existing multicast/unicast routing, with adaptation to changes in underlying unicast, multicast routes control protocol overhead to grow (at worst) linear in # receivers modular design for heterogeneous underlying technologies
RSVP: does not… specify how resources are to be reserved
rather: a mechanism for communicating needs
determine routes packets will take
that’s the job of routing protocols
signaling decoupled from routing
interact with forwarding of packets
separation of control (signaling) and data (forwarding) planes
Flow Specification Session must first declare its QoS
requirement and characterize the traffic it will send through the network R-spec defines the QoS being requested by receiver: Min Path Latency, Min Bdw, break bit, Hops, min MTU. T-spec defines the traffic characteristics of sender: Ps, rs, max burst size, min policed unit, max packt size. RSVP is the signaling protocol is needed to carry the R-spec and T-spec to the routers
Filter Specification The router needs to recognize the packets
belonging to that flow
IP of the sender IP destination Port number generating the packets Port number of the receiver Protocol ID Any field of the header
flowspec + filterspec = flowdescriptor
PATH Messages PATH messages carry sender’s Traffic
Specifications (TSpec) Carries also the FilterSpec Routers take note of the PATH sender and set up reverse path to it Receivers send RESV messages that follow reverse path and setup reservations If reservation cannot be made, user gets an error
RESV Messages RESV messages carry receiver’s QoS needs (R
spec) Forwarded via reverse path of PATH Queuing delay and bandwidth requirements Source traffic characteristics (from PATH) Filter specification Which transmissions can use the reserved resources? Reservation style. Router performs admission control and reserves resources
Router Handling of RESV Messages If new request rejected, send error message. If admitted: Install packet filter into forwarding dbase. Pass flow parameters to scheduler. Activate packet policing if needed. Forward RESV message upstream.
RSVP Functional Diagram Host
Router RSVPD
RSVPD Routing Process
Application D A T A
Packet Classifier
Policy Control
Policy Control
Admissions Control
Admissions Control
Packet Scheduler
DATA
Packet Classifier
Packet Scheduler
DATA
Soft State Routers keep state about reservation Periodic messages refresh state, with PATH
and RESV messages Non-refreshed state times out automatically Alternative: Hard state No periodic refresh messages. State is guaranteed to be there. State is kept till explicit removal.
Properties of soft state: Adapts to changes in routes, sources, and receivers. Recovers from failures Cleans up state after receivers drop out
RSVP Reservation (1)
R2
R3 PATH
2 1
PATH
R4 R1
3
Host A 24.1.70.210
1. An application on Host A creates a session, 128.32.32.69/4078, by communicating with the RSVP daemon on Host A. 2. The Host A RSVP daemon generates a PATH message that is sent to the next hop RSVP router, R1, in the direction of the session address, 128.32.32.69.
Host B 128.32.32.69 R5 3. The PATH message follows the next hop path through R5 and R4 until it gets to Host B. Each router on the path creates soft session state with the reservation parameters.
RSVP Reservation (2)
R2
R3 PATH
R4 PATH RESV
4 RESV
R1
Host A 24.1.70.210 4. An application on Host B communicates with the local RSVP daemon and asks for a reservation in session 128.32.32.69/4078. The daemon checks for and finds existing session state. 5. The Host B RSVP daemon generates a RESV message that is sent to the next hop RSVP router, R4, in the direction of the source address, 24.1.70.210.
5
Host B 128.32.32.69
6 R5 6. The RESV message continues to follow the next hop path through R5 and R1 until it gets to Host A. Each router on the path makes a resource reservation.
RSVP Multicast Reservation (1) Sender
PATH
R1
PATH
PATH
PATH
R2
R3
PATH
PATH
PATH
R4
R5 PATH
Receiver
R6
R7 PATH
RSVP Multicast Reservation (2) Sender
R1
R2
R4 Receiver
R3
R5
R6
R7
Reservation Merging
(3) 50Kbs (7) 100 Kbs R1
Reservations merge as they travel up tree.
(6) 100 Kbs R3
(2) 50Kbs (9) 60Kbs R4
(1) 50Kbs
Receiver #1
R6
(8) 60Kbs
Receiver #2
(5) 100 Kbs R7
(4) 100 Kbs
Receiver #3
RSVP: simple audio conference H1, H2, H3, H4, H5 both senders and receivers multicast group m1 no filtering: packets from any sender forwarded audio rate:
b
only one multicast routing tree possible H3
H2 R1
R2
H1 H5
R3
H4
RSVP: building up path state H1, …, H5 all send path messages on
m1:
(address=m1, Tspec=b, filter-spec=no-filter,refresh=100)
Suppose H1 sends first path message m1:
m1:
in L1 out L2 L6
in L7 out L3 L4
L6 m1: in out L5 L7
H3
H2
L3
L2
H1
L1
R1
L6
R2 L5
H5
L7
R3
L4
H4
RSVP: building up path state next, H5 sends path message, creating more state
in routers
m1:
L6 L1 m1: in out L1 L2 L6
in L7 out L3 L4
L5 L6 m1: in out L5 L6 L7
H3
H2
L3
L2
H1
L1
R1
L6
R2 L5
H5
L7
R3
L4
H4
RSVP: building up path state H2, H3, H5 send path msgs, completing path state
tables
m1:
L1 L2 L6 m1: in out L1 L2 L6
in L3 L4 L7 out L3 L4 L7
L5 L6 L7 m1: in out L5 L6 L7
H3
H2
L3
L2
H1
L1
R1
L6
R2 L5
H5
L7
R3
L4
H4
reservation msgs: receiver-to-network signaling reservation message contents: desired bandwidth filter type: • no filter: any packets address to multicast group can use reservation • fixed filter: only packets from specific set of senders can use reservation • dynamic filter: senders who’s packets can be forwarded across link will change (by receiver choce) over time. filter spec reservations flow upstream from receiver-to-senders,
reserving resources, creating additional, receiverrelated state at routers
RSVP: receiver reservation example 1 H1 wants to receive audio from all other senders H1 reservation msg flows uptree to sources H1 only reserves enough bandwidth for 1 audio stream reservation is of type “no filter” – any sender can use reserved bandwidth H3
H2
L3
L2
H1
L1
R1
L6
R2 L5
H5
L7
R3
L4
H4
RSVP: receiver reservation example 1 H1 reservation msgs flows uptree to sources routers, hosts reserve bandwidth b needed on
downstream links towards H1
m1: in L1 L2 out L1(b) L2
L6 L6
m1:
L2
H1
b b L1
R1
b L6
L7 L7(b)
L7 L6 L6(b) L7
m1: in L5 out L5
H2
L4 L4
in L3 out L3
b
R2 L5
H5
b L7
b
R3
L3 b L4
H3
H4
RSVP: receiver reservation example 1 (more) next, H2 makes no-filter reservation for bandwidth H2 forwards to R1, R1 forwards to H1 and R2 (?)
b already reserved on L6
R2 takes no action, since L6 m1: in L1 L2 out L1(b) L2(b) L6
m1:
b L2
H1
b b b L1
R1
b L6
L7 L7(b)
L7 L6 L6(b) L7
m1: in L5 out L5
H2
L4 L4
in L3 out L3
b
R2 L5
H5
b L7
b
R3
L3 b L4
H3
H4
b
RSVP: receiver reservation: issues What if multiple senders (e.g., H3, H4, H5) over link (e.g., L6)? arbitrary interleaving of packets L6 flow policed by leaky bucket: if H3+H4+H5 sending rate exceeds b, packet loss will occur L6 m1: in L1 L2 out L1(b) L2(b) L6
m1:
b L2
H1
b b b L1
R1
b L6
L7 L7(b)
L7 L6 L6(b) L7
m1: in L5 out L5
H2
L4 L4
in L3 out L3
b
R2 L5
H5
b L7
b
R3
L3 b L4
H3
H4
RSVP: example 2 H1, H4 are only senders send path messages as before, indicating filtered reservation Routers store upstream senders for each upstream link H2 will want to receive from H4 (only)
H3
H2
L3
L2
H1
L1
R1
L6
R2
L7
R3
L4
H4
RSVP: example 2 H1, H4 are only senders send path messages as before, indicating filtered reservation in
L1, L6 L2(H1-via-H1 out L6(H1-via-H1 L1(H4-via-R2
in
; H4-via-R2 ) )
)
L4, L7
L3(H4-via-H4 out L4(H1-via-R2 L7(H4-via-H4
; H1-via-R3 ) )
)
H3
H2
R2
L2
H1
L1
R1
L7
L6 in
L3
R3
L6, L7
L6(H4-via-R3 out L7(H1-via-R1
) )
L4
H4
RSVP: example 2 receiver H2 sends reservation message for source H4
at bandwidth b
propagated upstream towards H4, reserving b
in
L1, L6 L2(H1-via-H1 out L6(H1-via-H1 L1(H4-via-R2
H2 L2
H1
in
;H4-via-R2 (b)) ) )
L4, L7
L3(H4-via-H4 ; H1-via-R2 out L4(H1-via-62 ) L7(H4-via-H4 (b))
)
H3
b L1
R1
b L6 in
R2
b L7
L6, L7
L6(H4-via-R3 (b)) out L7(H1-via-R1 )
R3
L3 b L4
H4
RSVP: soft-state senders periodically resend path msgs to refresh (maintain) state
receivers periodically resend resv msgs to refresh (maintain) state path and resv msgs have TTL field, specifying refresh interval in
L1, L6 L2(H1-via-H1 out L6(H1-via-H1 L1(H4-via-R2
H2 L2
H1
in ;H4-via-R2 (b)) ) )
L4, L7
L3(H4-via-H4 ; H1-via-R3 out L4(H1-via-62 ) L7(H4-via-H4 (b))
)
H3
b L1
R1
b L6 in
R2
b L7
L6, L7
L6(H4-via-R3 (b)) out L7(H1-via-R1 )
R3
L3 b L4
H4
RSVP: soft-state suppose H4 (sender) leaves without performing teardown eventually state in routers will timeout and disappear!
in
L1, L6 L2(H1-via-H1 out L6(H1-via-H1 L1(H4-via-R2
H2 L2
H1
in
;H4-via-R2 (b)) ) )
L4, L7
L3(H4-via-H4 ; H1-via-R3 out L4(H1-via-62 ) L7(H4-via-H4 (b))
)
H3
b L1
R1
b L6 in
R2
b L7
L6, L7
L6(H4-via-R3 (b)) out L7(H1-via-R1 )
R3
L3 b L4
gone H4 fishing!
The many uses of reservation/path refresh recover from an earlier lost refresh message expected time until refresh received must be longer than timeout interval! (short timer interval desired)
Handle receiver/sender that goes away without
teardown
Sender/receiver state will timeout and disappear
Reservation refreshes will cause new reservations
to be made to a receiver from a sender who has joined since receivers last reservation refresh
E.g., in previous example, H1 is only receiver, H3 only sender. Path/reservation messages complete, data flows H4 joins as sender, nothing happens until H3 refreshes reservation, causing R3 to forward reservation to H4, which allocates bandwidth
RSVP scalability problems RSVP per-flow reservation model and soft-state philosophy are particularly suitable for multicast broadband applications (e.g. videoconference and video broadcasting) When used for point-to-point narrowband purposes (e.g. IP telephony) these choices implies large processing overhead in routers and great amount of traffic generation for periodic refreshes Example:
ADPCM coding requires 32 kb/s for a voice channel. Neglecting packet overhead, a single OC-12 interface of a backbone router (622 Mb/s) should support up to 20000 flows, implying that: • packet scheduler has to manage 20000 queues • up to 20000 states must be periodically refreshed
RSVP scalability problems Solutions: • Flows aggregation Protocol)
(SRP-
Scalable
Reservation
• Use of RSVP limited to the Access Network (IP based), with interconnection among IP domains relying on other QoS capable technologies (e.g. ATM) • Differentiated Services Approach • Combination of RSVP/IntServ in the Access Section and DiffServ in the backbone Internet
IETF Differentiated Services Concerns with Intserv: Scalability: signaling, maintaining per-flow router state difficult with large number of flows Flexible Service Models: Intserv has only two classes. Also want “qualitative” service classes
“behaves like a wire” relative service distinction: Platinum, Gold, Silver
Diffserv approach: simple functions in network core, relatively complex functions at edge routers (or hosts)
Diffserv Architecture Edge router:
r marking scheduling
per-flow traffic management marks packets as in-profile
and out-profile
Core router: per class traffic management buffering and scheduling based
on marking at edge preference given to in-profile packets
b
.. .
Edge-router Packet Marking profile: pre-negotiated rate A, bucket size B packet marking at edge based on per-flow profile
Rate A B
User packets
Possible usage of marking: class-based marking: packets of different classes marked
differently intra-class marking: conforming portion of flow marked differently than non-conforming one
Classification and Conditioning Packet is marked in the Type of Service (TOS) in
IPv4, and Traffic Class in IPv6 6 bits used for Differentiated Service Code Point (DSCP) and determine PHB that the packet will receive 2 bits are currently unused
Classification and Conditioning may be desirable to limit traffic injection rate of some class: user declares traffic profile (e.g., rate, burst size) traffic metered, shaped if non-conforming
Core routers forwarding Routers define packet classes and separate
incoming packets into classes. Treatment is done per class. Per-hop behavior (PHB) defines differences in performance among classes. PHB results in a different observable (measurable) forwarding performance behavior PHB does not specify what mechanisms to use to ensure required PHB performance behavior
Forwarding (PHB) PHBs developed: Expedited Forwarding: pkt departure rate of a
class equals or exceeds specified rate
logical link with a minimum guaranteed rate Providing low loss, low latency, low jitter, assured bandwidth, end-to-end service through DS domains Implies isolation: guarantee for the EF traffic should not be influenced by the other traffic classes Non-conformant traffic is dropped or shaped. Possible service: providing a virtual wire
Assured Forwarding: 4 classes of traffic each guaranteed minimum amount of bandwidth each with three drop preference partitions
Forwarding (PHB) PHBs developed: Assured Forwarding: The intent is that it will be used to implement services that differ relative to each other (e.g., gold, silver,…). AF defines 4 classes with some bandwidth and buffers allocated to them. • Each guaranteed minimum amount of bandwidth • Each with three drop preference partitions
Within each class, there are three drop priorities, which affect which packets will get dropped first if there is congestion. Non-conformant traffic is remarked.
AF table The DSCP (6 bit) pattern is: xyzab0 xyz is the class: 001-class1 ; 010-class2 ; 011-class3 ; 100-class4 ab is the drop precedence: 01-low ; 10-medium ; 11-high Class
Class 1
Class 2
Class 3
Class 4
001010 (AF11) 001100 (AF12) 001110 (AF13)
010010 (AF21) 010100 (AF22) 010110 (AF23)
011010 (AF31) 011100 (AF32) 011110 (AF33)
100010 (AF41) 100100 (AF42) 100110 (AF43)
Drop precedence
Low Drop Medium Drop High Drop
Service A service describes the overall treatment
of a customer’s traffic within a DS domain.
Customers see services, not PHBs.
To support a service, many components
must work together:
Mapping of service to PHBs, traffic conditioning, network provisioning, PHB-based forwarding.
Services in the DiffServ architecture is
defined in the form of Service Level Agreement (SLA).
QoS Summary
The brute force approach has many supporters…
Conventional IP Networks & Routing Client networks are connected to backbone via
edge routers
LAN, PSTN, ADSL
Data packets are routed based on IP address
and other information in the header Functional components Forwarding
• responsible for actual forwarding across a router • consists of set of procedures to make forwarding decisions
Control • responsible for construction and maintenance of the forwarding table • consists of routing protocols such as OSPF, BGP and PIM
Need for Multiprotocol Label Switching (MPLS) Forwarding function of a conventional router
a capacity demanding procedure constitutes a bottle neck with increase in line speed MPLS simplifies forwarding function by taking a totally different approach by introducing a connection oriented mechanism inside the connectionless IP networks
IP Router Control:
MPLS Control:
IP Router Software
IP Router Software
Forwarding:
Forwarding:
Longest-match Lookup
Label Swapping
ATM Switch Control: ATM Forum Software
Forwarding: Label Swapping
Label Switching Decomposition of network layer routing into
control and forwarding components applicable Label switching forwarding component algorithm uses
forwarding table label carried in the packet
What is a Label ? Short fixed length entity
Label Switching •Have a friend go to B ahead of you selecting the appropriate path. At every road they reserve a lane just for you. At ever intersection they post a big sign that says for a given lane which way to turn and what new lane to take.
LANE#1
LANE#1 TURN RIGHT USE LANE#2
LANE#2
MPLS and ISO model 7 to 5
Applications TCP
PPP PPP
UDP
IP MPLS Frame
4
3 ATM (*) ATM
2
Physical (Optical - Electrical)
1
Ethernet Relay
MPLS Shim Header The Label (Shim Header) is represented as a sequence of
Label Stack Entry Each Label Stack Entry is coded by 4 bytes (32 bits) as described 20 Bits is reserved for the Label Identifier (also named Label) Label (20 bits)
Exp S (3 bits) (1 bit)
TTL (8bits)
Label : Label value (0 to 15 are reserved for special use) Exp : Experimental Use S : Bottom of Stack (set to 1 for the last entry in the label) TTL :Time To Live
Label Values 0 - 15 Reserved
LABEL
DESIGNATION
0
IPv4 Explicit Null
1
Router Alert
2
IPv6 Explicit Null
3
Implicit Null
4-14
Reserved for Future Use
15
OAM
16 - 220-1
Production Use
Forwarding Equivalence Class Label Edge Router (LER) if it resides at the edge of an MPLS network and Label Switching Router (LSR) if it resides in the core on an MPLS network.
An MPLS capable router is called
Forwarding Equivalence Class (FEC): A subset of packets that are
all treated the same way by MPLS capable routers. A packet is assigned to an FEC at the ingress of an MPLS domain A packet’s FEC can be determined by one or more of the following: Source and/or destination IP address Source and/or destination port number Protocol ID Differentiated services code point Incoming interface A particular PHB (scheduling and discard policy) can be defined for a given FEC
Forwarding Equivalence Classes LSR
LER
LSR
LER
LSP IP1
IP2
IP1
#L1
IP1
#L2
IP1
#L3
IP2
#L1
IP2
#L2
IP2
#L3
Packets are destined for different address prefixes, but can be mapped to common path
• FEC = “A subset of packets that are all treated the same way by a router” • The concept of FECs provides for a great deal of flexibility and scalability • In conventional routing, a packet is assigned to a FEC at each hop (i.e. L3 look-up), in MPLS it is only done once at the network ingress
IP1
IP2
MPLS Operation 1a. Routing protocols (e.g. OSPF-TE, IS-IS-TE) exchange reachability to destination networks 1b. Label Distribution Protocol (LDP) establishes label mappings to destination network
4. LER at egress removes label and delivers packet
IP
IP
2. Ingress LER receives packet and “label”s packets
3. LSR forwards packets using label swapping
MPLS Operation At ingress LER of an MPLS domain, an MPLS
header is inserted to a packet before the packet is forwarded
Label in the MPLS header encodes the packet’s FEC
At subsequent LSRs The label is used as an index into a forwarding table that specifies the next hop and a new label. The old label is replaced with the new label, and the packet is forwarded to the next hop. Egress LER strips the label and forwards the
packet to final destination based on the IP packet header
Label Switched Path For each FEC, a specific path called
Switched Path (LSP) is assigned
Label
The LSP is unidirectional
To set up an LSP, each LSR must Assign an incoming label to the LSP for the corresponding FEC • Labels have only local significance
Inform the upstream node of the assigned label Learn the label that the downstream node has assigned to the LSP
Need a label distribution protocol so that an LSR
can inform others of the label/FEC bindings it has made A forwarding table is constructed as the result of label distribution.
LSP Route Selection Hop-by-hop routing: use the route determined by
the dynamic routing protocol Explicit routing (ER): the sender LSR can specify an explicit route for the LSP
Explicit route can be selected ahead of time or dynamically Advantages • Can establish LSP’s based on policy, QoS, etc. • Can have pre-established LSP’s that can be used in case of failures.
Signaling protocols • CR-LDP • RSVP-TE
MPLS BUILT ON STANDARD IP Dest 47.1 47.2 47.3
Dest 47.1 47.2 47.3
47.3 3
Out 1 2 3
3 1
Dest 47.1 47.2 47.3
Out 1 2 3
1
3
Out 1 2 3
1 47.1 2
2 47.2
2
• Destination based forwarding tables as built by OSPF, IS-IS, RIP, etc.
IP FORWARDING USED BY HOPBY-HOP CONTROL Dest 47.1 47.2 47.3
Dest 47.1 47.2 47.3
47.3 3 IP 47.1.1.1
Out 1 2 3
Out 1 2 3
3 IP 47.1.1.1 1 2
Dest 47.1 47.2 47.3
1
IP 47.1.1.1
Out 1 2 3
1 47.1 2 IP 47.1.1.1
2 47.2
Label Distribution Intf In 3
Intf Label Intf Label In In Out Out 3 50 1 40
Label Intf In Out 40 1 1
Request: 47.1
Dest Intf Label Out Out 47.1 1 50 47.3 3
3 2
3 1
47.1
1 2
Mapping: 40 47.2
2
Label Switched Path (LSP) Intf Label Dest Intf Label In In Out Out 3 0.50 47.1 1 0.40
3
Intf Dest Intf Label In Out Out 3 47.1 1 0.50
47.3 3 IP 47.1.1.1
1 2
Intf In 3
3
1 2
Label Dest Intf In Out 0.40 47.1 1
IP 47.1.1.1 1 47.1 2 47.2
EXPLICITLY ROUTED LSP ER-LSP
Intf Label Dest Intf Label In In Out Out 3 0.50 47.1 1 0.40 Intf In 3 3
Dest 47.1.1 47.1
47.3 3 IP 47.1.1.1
Intf Out 2 1
Label Out 1.33 0.50
3
1 2
Intf In 3
3
1 2
Label Dest Intf In Out 0.40 47.1 1
IP 47.1.1.1 1 47.1 2 47.2
Label Stacking A packet may carry multiple labels, organized as a
last-in-first-out stack A label may be added to/removed from the stack at any LSR Processing always done on the top label Allow the aggregation of LSPs into a single LSP for a portion of the route, creating a tunnel
It allows LSPs to be tunneled in other LSPs At the beginning of the tunnel, the LSR assigns the same label to packets from different LSPs by pushing the label onto each packet’s stack At the end of the tunnel, the LSR pops the top label
Label Stacking
Implicit Null
Explicit Null
