Internet Technology Voice over IP Peter Gradwell www.gradwell.com - - PowerPoint PPT Presentation

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Internet Technology Voice over IP Peter Gradwell www.gradwell.com - - PowerPoint PPT Presentation

Internet Technology Voice over IP Peter Gradwell www.gradwell.com | 01225 800 800 | info@gradwell.com BT Advert from 1980s http://www.youtube.com/v/o0H65_pag04 www.gradwell.com | 01225 800 800 | info@gradwell.com Page | 2 Welcome to


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Internet Technology – Voice over IP

Peter Gradwell

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BT Advert from 1980s

http://www.youtube.com/v/o0H65_pag04

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Welcome to Gradwell

Gradwell provides technology for every line on your business card Every product we sell solves a problem we had

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The TCP/IP Stack

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Where does VoIP fit in?

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Different Types of VoIP

VoIP Comes in Many Flavours

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Voice Protocols

  • All systems use some kind of application protocol to communicate
  • SIP
  • H323
  • IAX (Inter Asterisk eXchange)‏
  • Skinny (Cisco Protocol)‏
  • Audio is transported as RTP
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TCP/UDP/Ports

  • Traffic is either TCP or UDP
  • MTU on Ethernet typically 1500 bytes
  • 100 bytes of headers, 1400 bytes of data
  • Lots of packet overhead

MTU = Max Transfer Unit/A Packet Socket = IP Address + Port

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Good book:

  • If you're interested in more details
  • You can read most of it on amazon
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H.323

  • Point to Point Protocol
  • H.323 is actually a family of protocols
  • H.323 is used a lot on trunk routes, as it's signalling reflects ISDN, but is

becoming obsolete

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Modem over IP Conferencing Fax

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Inter-Asterisk eXchange

  • Asterisk is a major OpenSource phone system
  • IAX was invented by Mark Spencer/Digium
  • Single UDP stream – port 4569
  • Handles the Media inband on the same socket
  • Can support multiple RTP streams
  • Single signalling link
  • Far more bandwidth efficient
  • Also much friendlier for firewalls
  • Pronounced‏“eeks”

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SIP – Session Initiation Protocol

  • SIP is the dominant VoIP protocol
  • It's a peer to peer protocol, unlike H.323

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SIP Addressing

  • SIP is very similar to Email and HTTP
  • URI Address format:
  • sip:peter@gradwell.com
  • sip:100@sip.gradwell.net:5060
  • sip:441225800800@sip.gradwell.net

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Types of SIP Message

  • INVITE—An INVITE method indicates that the recipient user or service is invited to participate in a

session.

  • ACK - An ACK request confirms that the UAC has received the final response to an INVITE request.
  • ACK is used only with INVITE requests. ACK is sent end to end for a 200 OK response.
  • OPTIONS—A UA uses the OPTIONS request to query a UAS about its capabilities
  • BYE—A UA uses BYE to request the termination of a previously established session.
  • CANCEL—The CANCEL request enables UACs and network servers to cancel an in-progress

request, such as INVITE. This does not affect completed requests in which the UAS had already sent final responses.

  • REGISTER—A client uses a REGISTER request to register its current location information

corresponding to the AOR of the user with SIP servers.

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SIP Registration

  • Phone registers periodically to tell the SIP server where it is

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SIP Call Flow

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A Complete Call

16 Often we'll draw SIP call path diagrams out like this for debugging

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Debug using wireshark

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You can also do redirects

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Codecs

  • Audio is sampled at 8k or 16k times per second

(Hertz) – Pulse Code Modulation

  • Codecs trade off quality, bandwidth and CPU to

produce an RTP data packet

  • Narrow Band (8k) or Wide Band (16k)‏
  • We waste loads of Bandwidth on packet
  • verhead
  • Audio transmitted as RTP

Codec BR NEB G.711 64 Kbps 87.2 Kbps G.729 8 Kbps 31.2 Kbps G.723.1 6.4 Kbps 21.9 Kbps G.723.1 5.3 Kbps 20.8 Kbps G.726 32 Kbps 55.2 Kbps G.726 24 Kbps 47.2 Kbps G.728 16 Kbps 31.5 Kbps iLBC 15 Kbps 27.7 Kbps BR = Bit rate NEB = Nominal Ethernet Bandwidth (one direction)‏ Skype uses iLBC and ISAC

“Nyquist theorem states that if you sample an analog signal at twice the rate of the highest frequency of interest, you can accurately reconstruct that signal back into its analog form”

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Jitter & Packet Loss

  • ―Jitter‏is‏the‏variation‏of‏packet‏interarrival time‖
  • Jitter is a major problem
  • The internet is asymmetrical
  • Manage the problem with a Jitter buffer ~ 20ms
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Quality of Service

  • QoS is needed if a network link is congested
  • Routers need to enforce traffic prioritisation across

their links

  • How do they decide?
  • Trust the user? - DiffServ, Expedited Forwarding and

Low Delay bits in IP header

  • Trust the network – all traffic on port X or from IP

address Y is special

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Shift Change in the handset industry

What about Mobile Handsets?

  • SIP over WiFi has a very bad user

experience – Still difficult to setup

  • GSM‏Calls‏“Just‏work”
  • Costs of calling can be fixed

commercially – e.g. All inclusive call plans

  • Operators are not keen on offloading

voice minutes onto their data networks

  • iPhone & Google Voice are changing

things

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Which Mobile App Should I run?

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Google Voice Video

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http://www.youtube.com/v/m4Q9MJdT5Ds

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What is ENUM

  • User ENUM
  • Allows users to define how calls should be routed to their numbers
  • Stores telephone numbers in DNS and routing records
  • ENUM = E.164 NUmber Mapping (E164 is an ITU standard)
  • Needs‏a‏new‏‗root‘
  • .e164.arpa
  • Managed by RIPE
  • Numbers stored in reverse
  • +44 1865 332211 becomes
  • 1.1.2.2.3.3.5.6.8.1.4.4.e164.arpa
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What is ENUM

  • DNS uses a special form of record
  • Name Authority Pointer (NAPTR)
  • RFC2915
  • NAPTR contains a URI
  • Information on how to route call
  • Can specify multiple ways to call
  • VoIP (SIP/H323)
  • Email
  • Tel(ephone)
  • Fax
  • NAPTR records can be very sophisticated
  • Can contain time of day and other information
  • Can contain regular expressions so say a SIP record relates to the ENUM data
  • (Generic ENUM records for a telco)
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What is ENUM

  • This is a record for +1 770 555 1212
  • 100 / 102 is priority
  • Lowest number, highest priority
  • 10 is a preference level
  • Can be ignored here as both records 10
  • ―u‖‏is‏a‏user‏record‏i.e.‏stop‏processing
  • E2U+sip‏is‏the‏way‏of‏writing‏it‘s‏a‏SIP‏record
  • Regular expression and the final destination

$ORIGIN 2.1.2.1.5.5.5.0.7.7.1.e164.arpa. IN NAPTR 100 10 "u" "E2U+sip" "!^.*$!sip:information@pbx.example.com!i" . IN NAPTR 102 10 "u" "E2U+email" "!^.*$!mailto:information@example.com!i"

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ENUM call

+44 123 456789

SIP:01@A.co.uk

Company A

SIP:25@compB.co.uk

Company B

+44 987 654321

Dials +44 987 654321

Internet PSTN

No ENUM

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ENUM call

+44 123 456789

SIP:01@A.co.uk

Company A

SIP:25@compB.co.uk

Company B

+44 987 654321

Dials +44 987 654321

Internet PSTN

ENUM

+44 987 654321 SIP:25@compB.co.uk

+1.2.3.4.5.6.7.8.9.44.164.arpa

D N S

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ENUM in production

  • Asterisk‏supports‏ENUM‏―out‏the‏box‖
  • OpenSIPS has an ENUM module
  • Various PBX suppliers can support ENUM though may require special

configuration

  • Nominet have developed an Android application Enumdroid
  • Integrated into dialer
  • ENUM in the UK is currently looking for users
  • www.nominet.org.uk/enum/
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Enumdroid – real life app

  • Developed by Nominet’s Ray Bellis

– published as open source. – Available on the Android Marketplace

  • Dial the number of your contact
  • Application performs an ENUM

query and Presents the user the

  • ptions available
  • Caller make decision of call method

and places the call.

  • Presents a number of options

– Has flexibility to turn off GSM if necessary

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Enumdroid – real life app

  • Developed by Nominet’s Ray Bellis

– published as open source. – Available on the Android Marketplace

  • Dial the number of your contact
  • Application performs an ENUM

query and Presents the user the

  • ptions available
  • Caller make decision of call method

and places the call.

  • Presents a number of options

– Has flexibility to turn off GSM if necessary

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www.gradwell.com | 01225 800 800 | info@gradwell.com

Enumdroid – real life app

  • Developed by Nominet’s Ray Bellis

– published as open source. – Available on the Android Marketplace

  • Dial the number of your contact
  • Application performs an ENUM

query and Presents the user the

  • ptions available
  • Caller make decision of call method

and places the call.

  • Presents a number of options

– Has flexibility to turn off GSM if necessary

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www.gradwell.com | 01225 800 800 | info@gradwell.com

Enumdroid – real life app

  • Developed by Nominet’s Ray Bellis

– published as open source. – Available on the Android Marketplace

  • Dial the number of your contact
  • Application performs an ENUM

query and Presents the user the

  • ptions available
  • Caller make decision of call method

and places the call.

  • Presents a number of options

– Has flexibility to turn off GSM if necessary

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ENUM Conclusions

Story so far

  • ENUM allows interoperability between

carriers

  • More than just voice
  • Most applications can be completed

without‏“ENUM”,‏if‏you‏control‏the‏ network & devices

  • It needs a lot of end user applications
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Voice Over IP?

  • Protocols and services are well established & here to stay
  • Lines (no pun intended) are bluring
  • Network convergence is actually happening
  • No one wants a desk phone
  • Some of the mobile problems will be fixed commercially
  • Some problems by changes in end user behaviour and open access to

networks

  • The desktop environment is quickly unifying
  • Telcos still like building islands of subscribers
  • Regulator is looking at mobile number portability
  • ENUM gives end-user control, but no applications or drive for it

Where are we at with VoIP?

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Peter Gradwell T: 01225 800 810 E: peter@gradwell.com

Any Questions?