SIP, SDP and VoIP

1
SIP, SDP and VoIP

• David A. Bryan
• CSCI 434/534
• December 6, 2003

2
VoIP

• VoIP Voice over IP (Internet Protocol)
• Basic idea is to run telephone type services (and
  sometimes video) over IP
• Cost savings (administration, arbitrage)
• Tax/tariff avoidance
• Security (internal, encryption)
• Advanced features
• Next generation (3G and later) cellular phones
  run over IP networks

3
Telephony Protocols

• Original phone network is circuit switched
• SS7 is the protocol used for this NOT a
  computer standard
• Defined by ITU International Telecommunication
  Union
• VoIP is packet switched uses computer networks
• H.323 is also an ITU standard, but designed to
  work over packet switched networks

4
Telephony Protocols

• SIP
• IETF defined a protocol for VoIP, called SIP
• Designed for packet networks, useful for any sort
  of multimedia session establishment
• Support for integration with email/web services
• SIMPLE is a set of extensions to SIP to support
  instant messaging

5
H.323 vs. SIP
6
SIP Call
SIP/SDP
SIP/SDP
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SIP

• Session Initiation Protocol
• Defined (mostly) in IETF RFC 3261
• Many other drafts extend SIP and add new features
• Text based, very similar to HTTP
• Sets up a multimedia session between endpoints
• Allows mobility locate users using a URL
  (sipuser_at_sip.cs.wm.edu)
• Usually runs on port 5060
• Can use either UDP or TCP as transport
• Allows for IPv4 and IPv6. Next generation
  cellular phones (Europe) use SIP over TCP/IPv6

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SDP

• Session Description Protocol
• SIP doesnt try to describe the media stream
• Defined in IETF RFC 2327
• This is used by SIP to actually describe the type
  of media being used, where to send it, etc.
• Included in SIP message as a payload
• This is an older IETF protocol
• IETF tries to promote reuse whenever possible
• Yet another protocol, RTP, is used for the actual
  media

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SIP

• Logically, there are several parts to a SIP
  system
• UA (User-Agent, Endpoint) the phone itself
  can also be a softphone computer software
• Registrar keeps track of where the user is
  within a system
• Redirect Server used to inform devices when
  they need to contact different locations
• Proxy Server used to relay messages back and
  forth within the system
• In practice, several functions may actually be in
  the same server/program

10
Registration
When I get a call for Alice, Ill know to contact
her at 192.168.0.1!
Ah, now I will have to use 10.0.1.1 to reach
Alice.
11
Proxy
Bob is over at quickmart.com. Ill proxy the call
over there!
12
Redirection
13
SIP Messages

• SIP messages fall into two types requests and
  responses
• Requests include a method telling what it is they
  are doing.
• REGISTER, to register a phone or device
• INVITE, to start a new session (call)
• BYE, to end a session that is established
• CANCEL, to end a session that is still being set
  up (no final response yet)
• ACK, part of the triple handshake

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SIP Messages

• Responses are numeric, much like HTTP
• 1xx Provisional 100 Trying, 180 Ringing
• 2xx Successful 200 OK
• 3xx Redirection 301 Moved Permanently, 302
  Moved Temporarily
• 4xx Failure 404 Not Found, 410 Gone, 403
  Forbidden
• 5xx Server Failure 503 Service Unavailable
• 6xx Global Failure 600 Busy Everywhere

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Registration
16
Basic Call
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No answer
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INVITE message
INVITE sipbob_at_quickmart.com SIP/2.0 Via
SIP/2.0/TCP aliceua.phonecompany.com5060 From
Alice ltsipalice_at_phonecompany.comgt To Bob
ltsipbob_at_quickmart.comgt Call-ID
3848276298220188511_at_phonecompany.com CSeq 1
INVITE Contact ltsipalice_at_aliceua.phonecompany.c
omtransporttcpgt Content-Type application/sdp
Content-Length 151 v0 oCisco-UserAgent
10010 605 IN IP4 aliceua.phonecompany.com sSIP
Call cIN IP4 192.0.1.1 t0 0 maudio 49172
RTP/AVP 0 8 16
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INVITE message
INVITE sipbob_at_quickmart.com SIP/2.0 SIP
request, method is INVITE this message will
start a new call Via SIP/2.0/TCP
aliceua.phonecompany.com5060 Via header tracks
where it has been so far just Alices UA From
Alice ltsipalice_at_phonecompany.comgt Message is
from Alice, and this is the SIP URL for Alice To
Bob ltsipbob_at_quickmart.comgt Alice is trying to
contact Bob at his SIP URL Call-ID
3848276298220188511_at_phonecompany.com Call-ID is
a (hopefully) unique ID to track this particular
call. Alices UA generated this CSeq 1 INVITE
We also count messages within a call to see
what responses go with what messages. This first
exchange is an INVITE exchange. The response to
this will also have a CSeq of 1 INVITE Contact
ltsipalice_at_aliceua.phonecompany.comtransporttcpgt
How to contact Alice directly you can
sometimes skip the proxy. ltsnip!gt
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INVITE message
ltsnip!gt Content-Type application/sdp
Content-Length 151 The content of the message
is SDP to describe the media of the call. We
have 151 bytes of SDP v0 v is the SDP version
number -- 0 oCisco-UserAgent 10010 605 IN IP4
aliceua.phonecompany.com o is a description of
the UA and a session ID sSIP Call s is an
string to encode a name for this session cIN IP4
192.0.1.1 c tells the other side what address
to send the media to t0 0 t indicates start
and end times. 0 and 0 dont specify times in
advance maudio 49172 RTP/AVP 0 8 16 m describes
the media. We are listening for audio of type
RTP/AVP on port 49172
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Network Architecture

• Several ways to have a device that connects to an
  IP network
• True IP endpoint a phone w/an ethernet
  connection often quite expensive
• ATA Analog Telephone Adaptor (this is what you
  get if you sign up with Vonage) plugs into the
  ethernet, and you plug a phone into it
• IP enabled PBX (make an existing office phone
  system speak SIP)
• Soft Client program that runs on a computer and
  is used with microphone and speakers

22
Network Architecture

• How about connecting to the outside world?
• If you want to talk to another SIP user, your
  proxy may know how to contact that proxy directly
• To call out (or to have calls come in) you use a
  gateway. It connects to one or more regular phone
  lines to let you receive and place calls to old
  fashioned Plain Old Telephone Service, or POTS

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Gateway example
Gateway
SIP/IP network
POTS network
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Problems with VoIP

• Cost of entrance
• while the maintenance and operating costs are
  low, equipment can be expensive
• Reliability
• Still issues with reliability of systems
• Safety/Security
• 911 is a large issue, particularly routing to the
  right place
• Interception is difficult, and encryption is easy
  with softphones possibly a national security
  issue

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Problems with VoIP

• NAT and firewall traversal
• Media and signaling on different ports leads to
  issues with media not being able traverse
  firewalls
• Imbedded addresses within SIP messages are a
  problem for NATs
• Taxes/regulation
• Remove some incentive for VoIP

26
Open Source SIP projects

• Several good open source SIP projects
• VOCAL
• Full SIP softswitch includes proxy,
  softclient, voice mail, etc.
• ReSIProcate
• Newer, much nicer stack than VOCAL, but few
  applications so far
• IPTel Sip Express Router
• High performance Proxy. Less full featured than
  VOCAL, but faster and more robust

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References

• 1 SIP versus H.323, iptel.org,
  (http//www.iptel.org/info/trends/sip.html)
• 2 SIP Tutorial, iptel.org, (http//www.iptel.org
  /sip/siptutorial.pdf)
• 3 SIP RFC, RFC-3261, IETF, (http//www.ietf.org/
  rfc/rfc3261.txt)
• 4 SDP RFC, RFC-2327, IETF, (http//www.ietf.org/
  rfc/rfc2327.txt)
• 5 VOCAL Project, (http//www.vovida.org)
• 6 ReSIProcate Project, (http//www.resiprocate.o
  rg)
• 7 Vonage, (http//www.vonage.com)