Video

1
Video Voice over IP Communications A
SIP-based example

• Barry Andrews
• andrews_at_8x8.com

2
Why SIP?

• SIP (Session Initiation Protocol) is a
  communications protocol that defines how devices
  (computers, telephones, mobile phones) exchange
  information with each other.
• IETF (Internet Engineering Task Force) standard
  (RFC 3261-5, obsoletes RFC 2543 which
  originated in the mid-90s).
• Control Over Services is Pushed Out to the
  Endpoints
• In the traditional telecom environment,
  centralized switching elements control voice and
  other services
• By moving service control out to the endpoints
  (such as SIP-based mobile phones or audio
  clients), SIP eliminates the need for a central
  switching element.
• Flexible
• While SIP can set up any session (voice, video,
  messaging, games) it does not predefine what that
  session should be.
• It supports any MIME type in the messages and
  hence can support a broad set of services
  including new applications yet to be defined.
• Extensible
• SIP can be extended to support new messages and
  new types of services. E.g.. SIP for Instant
  Messaging and Presence (SIMPLE) is a SIP
  extension to support interoperable instant
  messaging and presence systems.
• Backwards compatible through base protocol

3
Why SIP?

• Integration with Internet Standards
• SIP uses URIs (Universal Resource Indicators),
  DNS (Domain Name Server), MIME ( Multipurpose
  Internet Mail Extensions) in ways that are
  compatible with other IP applications.
• Easily interoperates with Web applications
• SIP for Mobile Networks
• SIP chosen as the call control standard for 3G
  networks by 3GPP and 3GPP2
• IETFs SIP working groups looking at supporting
  3G SIP related requirements
• SIP for Wireline Networks
• Significant VoIP network deployments based on SIP
• International Softswitch Consortium (ISC) chose
  SIP as the signaling mechanism between
  softswitches.
• SIP used to provide Microsoft real-time
  communications capabilities in Windows Messenger
  and MSN Messenger clients
• SIP for Presence
• Instant Messaging (IM) and presence services
  based on proprietary technology (AOL, Yahoo!, and
  MSN)
• SIMPLE (SIP for Instant Messaging and Presence
  Leveraging Extensions) hopes to standardize
  instant messaging and presence among various
  service providers.

4
Packet8 Video Voice Solutions

• What is Packet8?
• A SIP based communication service
• Call to/from any phone anywhere in the world
• Plug and play installation. Dial tone in
  seconds.
• Multiple types of devices are supported
• IP phones
• Terminal adapters/analog telephones
• Windows Messenger
• IP videophones
• Choose your own area code - dedicated phone
  number associated with account
• NAT traversal
• Caller ID
• Call forwarding
• Voice and video mail (video mail in development)
• Free calls between Packet8 users
• Flat monthly service fee includes all features
  and bundled long distance minutes
• Portability
• Plug the phone into an Internet connection
  anywhere in the world to make and receive calls
• Receive calls on your Packet8 device or Windows
  Messenger

5
Packet8 Servlets

Traverse
Recorder
InCDR
InCPL
InCisco
Front
Player
Registrar
Gateway
Endpoints
OutCDR
Dialplan
OutCPL
OutCisco
Registration
Unsupported911
MyIP
MWI

• A SIP Servlet is a piece of code that can
  initiate, terminate or route SIP transactions and
  sessions. A complete application like Packet8 is
  built by assembling multiple servlets and
  VoiceXML scripts.

6
Packet8 Servlets

• Front Servlet
• Receives and sends all the SIP messages to and
  from the Packet8 endpoints.
• Its primary task is to enable SIP endpoints to be
  used behind a NAT.
• Its secondary task is to dispatch requests to the
  Registrar, Traverse, MWI or InCDR servlet.
• Registrar Servlet
• Used to store the registrations (mapping between
  an endpoint and its IP address) in a shared
  database.
• This database is also used by the InCPL servlet
  to find the destinations for an incoming call.
• Traverse Servlet
• Implements a traverse server and an RTP switch.
• The traverse server implements an 8x8 proprietary
  protocol to create an RTP switch.
• The RTP switch forwards the received RTP packets
  to a destination provided by the traverse
  protocol. This is used to enable RTP stream NAT
  traversal.
• Uses soft states, which means that the endpoint
  requesting a switch port must renew its request
  periodically. If an endpoint is brutally
  disconnected, the resources will be automatically
  de-allocated after a timeout delay.
• MWI Servlet
• Implements the IETF Message Waiting Indicator
  draft.
• Each endpoints subscribes to this servlet. Uses
  soft states, so the resources will be
  de-allocated if the endpoint does not renew its
  subscription.
• When a the subscriber mailbox status changes, a
  notification is sent back to the endpoint to
  update its status.

7
Packet8 Servlets

• OutCDR Servlet
• Creates and updates the Call Detail Record (CDR)
  for each call originating from an endpoint.
• Stores the time at the beginning of the call
  (INVITE), when the call is accepted or rejected
  (ACK or 400-699) and when the call is
  disconnected (BYE).
• DialPlan Servlet
• Uses an international dialplan to quickly accept
  or reject an invalid phone number.
• The endpoints are configured to send the phone
  number entered by the end-user very quickly to
  the Packet8 server (1 second).
• The dialplan servlet evaluates if a number is
  complete, incomplete or invalid
• If the number is complete, the call is proxied to
  the OutCPL servlet.
• If the number is invalid, the call is rejected.
• OutCPL Servlet
• Manages the authorizations for outgoing calls
  (International, CallerID, etc..). This
  information is stored in the OpenLDAP database.
• This is where the calls are dispatched to their
  destination
• Calls to a Packet8 endpoint are proxied to the
  InCDR servlet.
• Calls to an internal service (phone numbers
  starting with 0120) are proxied to this service
  (Voicemail, Unsupported911, MyIP).
• Calls to a PSTN number are sent to the OutCisco
  servlet.

8
Packet8 Servlets

• OutCisco Servlet
• The Cisco AS5300 gateway does not implement the
  latest version of SIP.
• This servlet implements a SIP B2BUA (Back to Back
  User-Agent) which isolates the Cisco gateway from
  the Packet8 network, repairs some of the
  incompatibilities with Packet8.
• InCisco Servlet
• This servlet does the same thing that the
  OutCisco servlet does, but for incoming calls.
• InCDR Servlet
• Creates and updates the Call Detail Record (CDR)
  for each incoming call. An incoming call can come
  from a PSTN gateway, or from the OutCPL servlet
  in the case of a Packet8 to Packet8 call.
• Stores the time at the beginning of the call
  (INVITE), when the call is accepted or rejected
  (ACK or 400-699) and when the call is
  disconnected (BYE).
• InCPL Servlet
• Manages the destinations for incoming calls. The
  information is stored in an OpenLDAP database and
  in the Registrar in-memory shared database.
• A call to the voicemail is directed to the Player
  servlet.
• A call to an endpoint has three phases
• The list of endpoints attached to the subscriber
  account are simultaneously called.
• After a delay, the call is also proxied to the
  recorder servlet (Voicemail).
• After a delay the call is also proxied to an
  additional phone number (forwarding). If the
  phone number is a PSTN number, this branch is
  proxied to the OutCDR servlet.

9
Packet8 Servlets

• Registration Servlet
• This servlet uses VoiceXML to implement an IVR.
  This is the IVR called when a subscriber
  registers his endpoint for the first time.
• The IVR checks the 10 digit code in the LDAP
  database and creates the binding in the OpenLDAP
  server
• Recorder Servlet
• This servlet uses VoiceXML to implement the
  recorder part of the voicemail application. This
  servlet is called when an endpoint does not
  respond to an incoming call.
• The servlet stores the audio message in an SQL
  database and uses the MWI servlet to update the
  message waiting indicator on the subscriber's
  endpoint(s).
• Player Servlet
• This servlet uses VoiceXML to implement the
  player part of the voicemail application.
• The caller uses menus to play and delete the
  messages in the mailbox.
• The application is also used to manage the
  mailbox.
• MyIP Servlet
• This servlet uses VoiceXML to play the caller's
  IP address.

10
Video Call Establishment - SDP Negotiation

• Non-proprietary
• All SDP parameters used as defined in RFC 2327
• Caller specifies the call options in the INVITE
  message
• choices for audio and video codecs
• audio packetization rate (in ms)
• maximum video bitrate acceptable to the caller to
  receive
• Callee responds with final call parameters in the
  200 (answer) response to the INVITE
• final choice of audio and video codec
• maximum video bitrate acceptable to the callee to
  receive
• Video encode bitrate
• both parties encode using the minimum of the two
  maximum values accepted by each party
• represents raw video content
• does not include, but must take into account
  packet overhead (headers)
• RTP UDP IP PPP Ethernet DSL/ATM
  (Gray DSL)

11
Example SDP Negotiation

• Callers SDP (INVITE)
• v0
• o0403242565 0 0 IN IP4 207.82.37.49
• s-
• cIN IP4 207.82.37.49
• t0 0
• maudio 8208 RTP/AVP 4 0 8 lt- Caller can
  receive audio codecs G.723.1,G.711U,G.711A
• aptime30
• mvideo 8210 RTP/AVP 34 31 lt- Caller can
  receive video codecs H.263 and H.261
• artpmap34 H263/90000
• artpmap31 H261/90000
• bAS640 lt- Caller can
  receive up to 640kbps video bitrate
• Callees SDP (200 OK response to INVITE)
• v0
• o0403242509 0 0 IN IP4 207.82.37.103
• s-

12
In Call Video Control Commands

• Non-proprietary scheme, with Packet8 extensions
• Control commands sent using SIP INFO method (with
  Call-ID of INVITE)
• INFO method defined in RFC 2976
• Message content is XML
• Described in draft-levin-mmusic-xml-media-control
  -01 (March 2003)
• Packet8 uses a subset of the commands, and adds
  two proprietary commands
• Control commands targeted at the remote partys
  video encoder
• Picture_Fast_Update Standard command to request
  intraframe when local decoder detects error
• Bandwidth_Step_Up Packet8 extension to request
  remote encoder to use higher bit rate
• Bandwidth_Step_Down Packet8 extension to request
  remote encoder to use lower bit rate
• Controls may be issued manually (by user) or
  automatically
• keypad entry allows user to control bitrate
• decoder can make decisions based on perceived
  error rate and/or latency
• high decode error rates and/or high packet
  latency may require lower bitrate to resolve

13
Example INFO Command

• Example (Fast Update Request for intraframe)
• INFO sip0403242565_at_207.82.37.495060 SIP/2.0
• Call-ID 8076bb8-31f5d21-cf5238f5_at_packet8.net
• From ltsip0403242509_at_packet8.netgttag402fff4
• To "BVP8770 "ltsip0403242565_at_packet8.netgttag403
  1e7c-31f5d21
• CSeq 101 INFO
• Via SIP/2.0/UDP 207.82.37.485060branchz9hG4bK.
  4035e6c.31f510erport
• Contact BVP8770ltsip0403242509_at_207.82.37.485060gt
  
• Max-Forwards 70
• Route ltsip192.84.18.2545060lrgt
• User-Agent 8x8 SIP Phone
• Content-Type text/plain
• Content-Length 195
• lt?xml version"1.0" encoding"utf-8" ?gt
• ltmedia_controlgt
• ltvc_primitivegt
• ltto_encodergt

14
Packet8 Registration
The endpoint sends a REGISTER
Endpoint
Packet8
REGISTER
REGISTER sippacket8.net SIP/2.0 Via SIP/2.0/UDP
10.1.28.247branchz9hG4bK.28f98.676a9102rport Fr
om ltsip02104412_at_packet8.netgttagt286c0a676a9101
g To ltsip02104412_at_packet8.netgt Call-ID
4aecc-114c-0-a0130bb_at_packet8.net CSeq 64649
REGISTER Contact sip02104412_at_10.1.28.247 Expires
180 Max-Forwards 70 User-Agent DTA SIP/0.11.8
NNOS/VR30 Content-Length 0
15
Packet8 Registration
Packet8 stores the mapping
Endpoint
Packet8
REGISTER
200 OK
SIP/2.0 200 OK Via SIP/2.0/UDP
10.1.28.247branchz9hG4bK.28f98.676a9102rport50
60 received12.234.8.147 From
testltsip02104412_at_packet8.netgttagt286c0a676a9101
g To testltsip02104412_at_packet8.netgttag9zvw48shd
nbemkglcny5b9wuy Call-ID 4aecc-114c-0-a0130bb_at_pac
ket8.net CSeq 64649 REGISTER Server
Packet8/0.2.6 (RegistrarServlet)
eSLEE/0.8.9 Content-Length 0 Expires
30 Contact ltsip02104412_at_10.1.28.247gtexpires30
16
Message Waiting Indicator
The endpoint subscribes to the MWI events
Endpoint
Packet8
SUBSCRIBE
SUBSCRIBE sip02104412_at_packet8.net SIP/2.0 Via
SIP/2.0/UDP 10.1.28.247branchz9hG4bK.28c3c.192e
rport From testltsip02104412_at_packet8.netgttagt28
e6ca192eg41a7 To testltsip02104412_at_packet8.netgt C
all-ID 4b678-192d-a2f420-a0130bb_at_packet8.net CSeq
100 SUBSCRIBE Contact sip02104412_at_10.1.28.247
Expires 1800 Max-Forwards 70 Event
message-summary User-Agent DTA SIP/0.11.8
NNOS/VR30 Content-Length 0
17
Message Waiting Indicator
Packet8 accepts
Endpoint
Packet8
SUBSCRIBE
200 OK
SIP/2.0 200 OK Via SIP/2.0/UDP
10.1.28.247branchz9hG4bK.28c3c.192e
rport5060received12.234.8.147 From
testltsip02104412_at_packet8.netgttagt28e6ca192eg41a
7 To testltsip02104412_at_packet8.netgttagdimn9pu1s
xelbix6kr4xsl5s4 Call-ID 4b678-192d-a2f420-a0130b
b_at_packet8.net CSeq 100 SUBSCRIBE Record-Route
ltsip02104412_at_192.84.18.2515060lrgt Contact
ltsip02104412_at_192.84.18.2525066gt Server
eSLEE/0.8.7 Packet8-MWINotifier
(StVallier) Content-Length 0 Expires 1800
18
Message Waiting Indicator
And sends the current state
Endpoint
Packet8
SUBSCRIBE
200 OK
NOTIFY
NOTIFY sip02104412_at_10.1.28.247 SIP/2.0 Via
SIP/2.0/UDP 192.84.18.2515060
branchz9hG4bK9rp493qfntaagovhw7t0qixro Via
SIP/2.0/UDP 192.84.18.2525066
branchz9hG4bKbnsrkh0lc50bhwp8tv5qoe4s0 From
testltsip02104412_at_packet8.netgttagdimn9pu1sxelbix
6kr4xsl5s4 To testltsip02104412_at_packet8.netgttag
t28e6ca192eg41a7 Call-ID 4b678-192d-a2f420-a0130b
b_at_packet8.net CSeq 1964223776 NOTIFY Contact
ltsip02104412_at_192.84.18.2525066gt Max-Forwards
69 Event message-summary Subscription-State
active Content-Length 44 Content-Type
application/simple-message-summary User-Agent
eSLEE/0.8.7 Packet8-MWINotifier
(StVallier) Record-Route ltsip02104412_at_192.84.18.
2515060lrgt Message-Waiting yes Voicemail 9/9
(0/0)

19
Message Waiting Indicator
The endpoint accepts the current state
Endpoint
Packet8
SUBSCRIBE
200 OK
NOTIFY
200 OK
SIP/2.0 200 OK Via SIP/2.0/UDP
192.84.18.2515060 branchz9hG4bK9rp493qfntaago
vhw7t0qixro Via SIP/2.0/UDP 192.84.18.2525066
branchz9hG4bKbnsrkh0lc50bhwp8tv5qoe4s0 From
testltsip02104412_at_packet8.netgttagdimn9pu1sxelbix
6kr4xsl5s4 To testltsip02104412_at_packet8.netgttag
t28e6ca192eg41a7 Call-ID 4b678-192d-a2f420-a0130b
b_at_packet8.net CSeq 1964223776 NOTIFY Record-Route
ltsip02104412_at_192.84.18.2515060lrgt Server
DTA SIP/0.11.8 NNOS/VR30 Content-Length 0
20
Message Waiting Indicator
Later, Packet8 updates the MWI state
Endpoint
Packet8
SUBSCRIBE
200 OK
NOTIFY
NOTIFY sip02104412_at_10.1.28.247 SIP/2.0 Via
SIP/2.0/UDP 192.84.18.2515060
branchz9hG4bK9rp493qfntaagovhw7t0qixro Via
SIP/2.0/UDP 192.84.18.2525066
branchz9hG4bKbnsrkh0lc50bhwp8tv5qoe4s0 From
testltsip02104412_at_packet8.netgttagdimn9pu1sxelbix
6kr4xsl5s4 To testltsip02104412_at_packet8.netgttag
t28e6ca192eg41a7 Call-ID 4b678-192d-a2f420-a0130b
b_at_packet8.net CSeq 1964223777 NOTIFY Contact
ltsip02104412_at_192.84.18.2525066gt Max-Forwards
69 Event message-summary Subscription-State
active Content-Length 44 Content-Type
application/simple-message-summary User-Agent
eSLEE/0.8.7 Packet8-MWINotifier
(StVallier) Record-Route ltsip02104412_at_192.84.18.
2515060lrgt Message-Waiting yes Voicemail
10/10 (0/0)
200 OK
NOTIFY
21
Message Waiting Indicator
The endpoint accepts the update
Endpoint
Packet8
SUBSCRIBE
200 OK
NOTIFY
200 OK
NOTIFY
200 OK
SIP/2.0 200 OK Via SIP/2.0/UDP
192.84.18.2515060 branchz9hG4bK9rp493qfntaago
vhw7t0qixro Via SIP/2.0/UDP 192.84.18.2525066
branchz9hG4bKbnsrkh0lc50bhwp8tv5qoe4s0 From
testltsip02104412_at_packet8.netgttagdimn9pu1sxelbix
6kr4xsl5s4 To testltsip02104412_at_packet8.netgttag
t28e6ca192eg41a7 Call-ID 4b678-192d-a2f420-a0130b
b_at_packet8.net CSeq 1964223777 NOTIFY Record-Route
ltsip02104412_at_192.84.18.2515060lrgt Server
DTA SIP/0.11.8 NNOS/VR30 Content-Length 0
22
NAT Traversal Protocols

• Network Address Translators (NATs), while
  providing many benefits, also come with many
  drawbacks. The most troublesome of those
  drawbacks is the fact that they break many
  existing IP applications, and make it difficult
  to deploy new ones. Examples of such protocols
  include multimedia applications and file sharing.
• When the Packet8 Service was designed, there was
  one standard solution for NAT traversal - Simple
  Traversal of UDP Through NAT (STUN)
• STUN (RFC 3489) is a client server protocol used
  to detect what kind of NAT is used, and to
  discover the NAT mapping that can be used for
  sending the incoming RTP stream.
• STUN by itself does not provide a complete
  solution for NAT traversal. A complete solution
  requires a means by which a client can obtain a
  transport address from which it can receive media
  from any peer which can send packets to the
  public Internet. This can only be accomplished by
  relaying data though a server that resides on the
  public Internet.
• TURN (Traversal Using Relay NAT) is a new IETF
  draft that allows a client to obtain IP addresses
  and ports from such a relay.

23
NAT Traversal Protocols

• Although TURN will almost always provide
  connectivity to a client, it comes at high cost
  to the provider of the TURN server. It is
  therefore desirable to use TURN as a last resort
  only, preferring other mechanisms (such as STUN
  or direct connectivity) when possible.
• As TURN was not available when Packet8 was
  designed, we chose to design and implement our
  own protocol, which is described in the following
  slides but which is similar in principle to TURN.