Overview

• What is VoIP?
• Toll ByPass
Voice over IP - VoIP • RTP
• Control Protocols
• MGCP
SMD151
• H323
Peter.Parnes@ltu.se
• SIP in detail

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What is Voice over IP Categories of VoIP

• The packetisation and transport of classic
public switched telephone system audio
over an IP network.
• Private and Desktop: Skype, Microsoft Netmeeting,
ohphone, gphone, Asterisk*, Marratech, etc.
• Enterprise : Small IP phone deployments, IP PBX,
Cisco Callmanager.

• The analog audio stream is encoding in a
digital format, with possible compression,
and encapsulating it in IP for transport
over your LAN/WAN or the public internet
• ISP/Carrier : Toll Bypass services, VOIP wholesale,
using equipment by vendors such as Cisco, Lucent,
Avaya, etc.
• Telco Grade : Local service, Last mile delivery, total
phone services, high dependability and availability.

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Toll ByPass How does it work?

• VoIP is not a protocol.
• VoIP is a collection of protocols and devices that
allow for the encoding, transport and routing of
audio calls over IP networks.

• PSTN  VoIP  PSTN

• Native VoIP  PSTN

• Native VoIP  Native VoIP

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 The Guts RTP
• RTP (Real-Time Transport Protocol) • RTP data packet header
• RTCP (Real-Time Control Protocol)
• RTP is a UDP stream with no intelligence
for QoS or resource reservation
• Contains a packet number for detection of
packet loss and re-sequencing of out of
order packets.
• Unidirectional : two streams in any call

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RTCP Sender Report RTCP Receiver Report

• Real-time Transfer Control Protocol • Receiver report
Sender Report

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Control Protocols - SIP Control Protocols – MGCP

• SIP : Session Initiation Protocol.

• SIP is a textual based client/server protocol. This • MGCP : Media Gateway Control Protocol.
makes debugging easy as it’s invite etc
messages are human readable.
INVITE sip:021326001@203.109.130.94:5060;user=phone SIP/2.0 • MGCP is a master/slave protocol where all
Via: SIP/2.0/UDP 10.30.60.20:5060
From: "Mr Anderson" <sip:099639951@10.30.60.20:5060>;tag=1ACC.8D00 the smarts resides in the gateway
To: <sip:021326001@203.109.130.94:5060>;tag=359870F0-16CD
Call-ID: 00C0.95C9.5818.4013.1ACC.8D00@10.30.60.20 controller and not the gateway.
CSeq: 5257 ACK
Content-Length: 0
Max-Forwards: 70
Contact: sip:099639951@10.30.60.20:5060

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 Control Protocols – H323 Control Protocols – H323
value H323_UserInformation ::=
{
h323-uu-pdu
{
h323-message-body releaseComplete :
{
protocolIdentifier { 0 0 8 2250 0 4 }
callIdentifier
{
guid '3E940C894E1311D8A33E919F0987C365'H
}
}
h245Tunneling TRUE
}
}

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H323 H323 Call Setup Version 1

H.323 call setup consisted of:
• Q.931 call signaling (1st TCP connection)
• H.245 capability exchange (2nd TCP connection)
• H.225.0 establish logical channels (3rd TCP
connection)
• Now send RTP...
• Each TCP connection requires a SYN exchange
before data can be sent, plus the usual TCP
timeouts on packet loss.
• Connection setup was excessively slow.

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H323 Evolution H323 FastStart

• H.323 has evolved so that many of the
original defects have been reduced.
• Unfortunately you still need to implement the
old mechanisms too to be compliant.
• Two new methods of setup:
• H.245 tunneling over the H.225.0 connection.
• H.323v2 FastStart

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 Control Protocols Summary
• SIP • What is VoIP?
• MGCP • Toll ByPass
• H323 • RTP
• Skinny : Cisco’s IP phone control protocol • Control Protocols
• H323
• MGCP
• SIP

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Questions? SIP Overview

• SIP - Introduction
• SIP architecture and philosophy

?
• Methods Used in SIP
• SIP messages & responses
• Security
• Summary
• References
Nishita Vora Pandya
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SIP is SIP Basic Functionality

• An Application-layer control (signaling) protocol Supports 5 facets of communication:
• creating, modifying and terminating sessions with one or more
participants. • User location: determination of the end system
• Sessions include: to be used for communication;
• Internet multimedia conferences, • User capabilities: determination of the media
• Internet telephone calls and media parameters to be used;
• multimedia distribution.
• User availability: determination of the
• Members in a session can communicate via:
• Multicast
willingness of the called party to engage in
• mesh of unicast relations communications;
• combination of these. • Call setup: "ringing", establishment of call
• Text based parameters at both called and calling party;
• Model similar to HTTP : uses client-server model
• Call handling: including transfer and termination
of calls.
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 SIP Functionality (cont.) Development of SIP

• SIP can also initiate multi-party calls
using a multipoint control unit (MCU) or
fully-meshed interconnection instead of
multicast.
• You can try in Marratech…
• Internet telephony gateways that connect
Public Switched Telephone Network
(PSTN) parties can also use SIP to set up
calls between them.
• SIP developed by Handley, Schulzrinne, Schooler, and
Rosenberg
- Submitted as Internet-Draft 7/97
• Assigned RFC 2543 in 3/99
• Goals: Re-use of & Maximum Interoperability with
existing protocols
• Alternative to ITU’s H.323
- H.323 used for IP Telephony since 1994
- Problems: No new services, addressing, features
- Concerns: scalability, extensibility

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SIP Philosophy SIP Architecture

• Internet Standard • SIP uses client/server architecture
- IETF - http://www.ietf.org • Client - originates message
• Server - responds to or forwards message
• Reuse Internet addressing (URLs, DNS, proxies)
• Elements:
- Utilizes rich Internet feature set
• SIP User Agents (SIP Phones)
• Reuse HTTP coding • SIP Servers (Proxy or Redirect - used to locate SIP
- Text based users or to forward messages.)
• Makes no assumptions about underlying • Can be stateless or stateful

protocol: • SIP Gateways:

- TCP, UDP, X.25, frame, ATM, etc. • To PSTN for telephony interworking

- Support of multicast • To H.323 for IP Telephony interworking

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SIP Entities – User Agents SIP Entities – Network Servers

• User Agents • Network Servers
• User Agent Client (UAC) • Proxy: Decides next hop and forwards
• Initiates SIP requests request, relays call signaling , operates in a
transactional manner, saves no session state
• User Agent Server (UAS)
• Redirect: Sends address of next hop back to
• Returns SIP responses client, redirects callers to other servers
• Registrar: Accepts REGISTER requests from
clients, maintains users’ whereabouts at a
location server
• Diff. types may be co-located.

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 SIP Operation Step 1:SIP Addressing

1. SIP Addressing Uses Internet URLs

• Uniform Resource Identifiers
2. Locating a SIP Server • Supports both Internet and PSTN addresses
3. Sending SIP Requests : SIP • General form is name@domain
• To complete a call, needs to be resolved down to User@Host
Transactions • Examples:
4. SIP Methods sip:peter@parnes.com
sip:J.T. Kirk <kirk@starfleet.gov>
5. SIP Responses
sip:+46-8-50555699@digisip.net;user=phone
6. Subsequent Requests and Responses sip:guest@10.64.1.1

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Step 2: Locating a SIP server Locating (cont.)

• A caller first locates the appropriate server Client

1. Send the request to the server’s IP address if the host
• When client wants to send a request URI part of request URI is an IP address otherwise
2. Should try to contact a server at the port listed in
client will either send it to request URI. If no port specified then try port 5060
- Locally configured Proxy server 1. Use specified protocol if applicable
2. otherwise use UDP if supported,
- IP address & port corresponding to the 3. if UDP fails or otherwise use TCP
3. Find one or more address of server by querying DNS,
request URI [similar to the one in step 1] • NAPTR, SRV, CNAME, A
• Client must determine IP address, port of 4. Results MAY be cached.
server and the protocol to be used. 5. Capability to interpret ICMP messages must exist

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Step 3:Send a SIP request Order of Operation

• Once the host part has been resolved to a SIP server, - • Step 1: Caller Issues Initial INVITE Request
client sends 1 / more SIP requests to that server
• receives 1 / more responses from the server. • Step 2: Callee Issues Response
• SIP Request-line (Messages) defined as: • Step 3: Caller Receives Response to Initial
<Method> <SP> Request-URI <SP>SIP-Version
<CRLF> request
(SP=Space, CRLF=Carriage Return and Line Feed) • Step 4: Caller or Callee Generate Subsequent
(Method = “INVITE” | “ACK” | “OPTIONS” | requests
“BYE” | “CANCEL” | “REGISTER”)
• Example: • Step 5: Receive Subsequent Requests
INVITE sip:peter@parnes.com SIP/2.0 • Step 6: BYE to end session
• Step x: CANCEL may be issued

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 Simple Example Methods (cont.)
• INVITE: Initiates sessions
- Session description included in message body
INVITE
• Re-INVITEs used to change session state
200 OK
• ACK confirms session establishment, can only
ACK be used with INVITE
• BYE terminates a session (hanging up)
Audio, video, ...
• CANCEL cancels a pending invite
• REGISTER: binds a permanent address to
BYE
current location, may convey user data
• OPTIONS: capability inquiry
200 OK

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Redirect Example Redirect Example

INVITE
200 OK
•
•
•
REGISTER
INVITE
302 Redirect
200 OK

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Relaying a Call Relay Example

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 Relay and Topology Hiding SIP Negotiation
• Optimistic Call

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SIP Negotiation Multi-register Example

• Pessimistic Call
REGISTER

REGISTER

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Proxy Server Example Redirect Server Example

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 SIP Responses SIP Responses (cont.)

• 1xy – Informational
• SIP Responses defined as (HTTP-style):
request received , continuing to process request
SIP-Version SP Status-Code SP Reason-Phrase • 2xy – Success
CRLF action successfully recvd., understood & accepted
(SP=Space, CRLF=Carriage Return and Line Feed) • 3xy – Redirection
Further action to be taken to complete the request
• Example: • 4xy – Client error
SIP/2.0 404 Not Found request contains syntax error or cant be completed at this
server
• First digit gives Class of response • 5xy – Server error
server fails to fulfill an apparently valid request
• 6xy – global failure,
request is invalid at any server

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Step 4 Generate Subsequent Requests SIP Requests Example

• Once the call has been established Required Headers (fields):
• either the caller or callee may generate INVITE or BYE
requests to change or terminate the call. INVITE sip:picard@wcom.com SIP/2.0
Via: SIP/2.0/UDP host.wcom.com:5060
• For the desired call leg the headers are set as From: Alan Johnston <sip:alan.johnston@wcom.com>
follows (both including any tags): To: Jean Luc Picard <sip:picard@wcom.com>
- the To header field is set to the remote address, Call-ID: 314159@host.wcom.com
CSeq: 1 INVITE
and
- the From header field is set to the local address. • Via: Shows route taken by request.
• The Contact header field may be different than the • Call-ID: unique identifier generated by client.
Contact header field sent in a previous response or • CSeq: Command Sequence number
request. • generated by client
• The Request-URI may be set to the value of the Contact
• Incremented for each successive request
header field received in a previous request or response
from the remote party, or to the value of the remote
address. [supports mobility]
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Via Field in Header Via Headers and Routing

• The Request headers include a Via field
• The Via field indicates the path taken by the request
so far.
• Every proxy adds a Via Header with its address
to make sure that responses within a transaction
take the same path (to avoid loops, or to make
sure that same firewall will be hit on the way
back)
• This prevents request looping and ensures replies
take the same path as the requests, which assists in
firewall traversal and other unusual routing situations.
• Via headers are used for routing SIP messages
• Requests
• Request initiator puts address in Via header
• Servers check Via with sender’s address, then add own address,
then forward. (if different, add “received” parameter)
• Responses
• Response initiator copies request Via headers.
• Servers check Via with own address, then forward to next Via
address
• All Via headers are copied from request to response in
order
• Response is sent to address in top Via header

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 Via Header (cont.)
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Via Example Response
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Reliability
• If UDP is used:
-SIP client should retransmit a BYE, CANCEL,
OPTIONS, or REGISTER request, exponential backoff,
starting at a T1 second interval, doubling the interval for
each packet, and capping off at a T2 second interval.
-Retransmit a INVITE request with an interval that starts
at T1 seconds, exponential back off, cease
retransmissions if a provisional or definitive response
recvd., or once it has sent a total of 7 request packets
• Clients using TCP do not need to retransmit requests
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VIA Example
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Step 5 Receiving Subsequent Requests
• Subsequent to receipt, the following checks are made:
1. If the Call-ID is new,
- the request is for a new call, regardless of the values
of the To and From header fields.
2. If the Call-ID exists,
- the request is for an existing call.
- If the To, From, Call-ID, and CSeq values exactly match (including
tags) those of any requests received previously,
- the request is a retransmission.
3. If there was no match to the previous step,
- To & From fields compared against existing call leg local and
remote addresses.
- If there is a match, & the CSeq in the request > last CSeq
received on that leg,
- the request is a new transaction for an existing call leg.
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Authentication & Encryption
• SIP supports a variety of approaches:
• end to end encryption
• hop by hop encryption
• Proxies can require authentication:
• Responds to INVITEs with 407 Proxy-
Authentication Required
• Client INVITEs again with Proxy-Authorization
header.
• Stacked!!
• SIP Users can require authentication:
• Responds to INVITEs with 401 Unauthorized
• Client INVITEs with Authorization header
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 Features and Benefits
Feature Benefits
Internet SIP-based systems can take advantage of the
Enabled growth of the Internet. Translating gateways
permit SIP-based systems to contact parties on
the Public Switched Telephone Network
(PSTN) without being encumbered by its legacy
standards.
Scalability Architecture permits inexpensive scaling. H/W
& S/W requirements for adding new users to
SIP-based systems is greatly reduced.
Simplicity SIP stack is smaller. SIP can be considered as
a simple toolkit that enables smart endpoints,
gateways, processes & clients to be built and
implemented
Features and Benefits(cont.)
Feature Benefit
Distributed -De-centralized intelligence permits more
functionality functionality within each component.
-- Changes made to specific components have a
minor impact on the rest of the system. It is
possible to connect one SIP phone to another
with an Ethernet cable & make calls between the
sets without the aid of any other server modules.
- The other system components become useful
when the network requires more than two
phones.
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SIP vs. H323 SIP Summary

• H.323 uses TCP for setup • SIP is:

• UDP gives better timing control
• UDP is critical for very large proxies that wish to be
stateless.
• H.323 uses ASN.1 Packed Encoding Rules
• For a long time only one company sold an ASN.1
compiler that could compile H.323's ASN.1
• SIP uses an HTTP-like syntax.
• Much easier to write and to debug.
• Much experience with HTTP
• Relatively easy to implement
• Gaining vendor and carrier acceptance
• Very flexible in service creation
• Extensible and scaleable
• Appearing in products right now
• SIP provides its own reliability mechanism & is
therefore independent of the packet layer and
only requires an unreliable datagram service

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But … Questions?
• SIP is not:
•Going to make PSTN inter-working easy

?
•Going to solve all IP Telephony issues (QoS)
• Secure !?

• SIP is still evolving and being extended as

technology matures and SIP products are
socialized in the marketplace.
• Newer RFCs: #3261 through #3265

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