VOIP Technology (SIP, SDP, RTP...), SBC.....................

Telecom Testing
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Wireline Wireless
(Broadband,SS7.....) (GSM,CDMA,Wi-Fi,3G,4G.......technologies)

* VOIP(SIP) Technology supports both Wireline and Wireless.

VOIP(Voice Over Internet Protocol):

It is the internet technology to carry voice communication and

multimedia sessions over Internet protocol networks such as internet.

The steps involved in originating voip telephone call are signalling and
media channel setup,digitization of the analog voice signal, encoding
,packetization and Transmission as Internet protocol packets over a packet switched
network.

Voip systems employ session control protocols to control the set-up

and tear-down of calls.There are three types of Voip tools that are commonly used,
IP phones, Softphone and Mobile Softphone.

IP Phones: IP Phones are the most institutionally established but still the
least obvious of the VOIP tools.
Softphones: It�s use has increased during the global recession as many
persons, looking for ways to cut costs have turned to these tools for free or
inexpensive calling or video conferencing applications.

Three classes of software IP: Web calling,voice and video instant

messaging and web conferencing.

Mobile Softphone: It is just another example of the adaptability of voip.

Voip is available on may smart phones and internet devices.

Voip has been implemented in various ways using both proprietary

and open protocols and standards.

PROTOCOLS:
� H.323
� Ip multimedia subsysem (IMS)
� Media Gateway Control Protocol (MGCP)
� Session Initiation protocol (SIP)
� Real-Time transport protocal (RTP)
� Session Description Protocol (SDP)
The H.323 protocol was one of the, first viop protocol that found wide spread
implementation for long-distance traffic as well as local area networks. Since
development of newer, less complex protocols , such as MGCP and SIP. The sip has
gained wide spread voip market penetration.

� IETF Standard: Sip, Mgcp .

� ITU Standard : H.323 ,Megaco .
� Proprietory : IAX(Internation Asterix) , Sccp(Cisco) .
SIP and H.323 are signalling protocols.
Mgcp and Megaco are Media Gateway protocol
Voip Advantages:

1) Cheap calls or Free calls.

2) Low Equipment Cost .
3) Voice,Data,Video and cable,Iptv.
4) Qos (Quality of service).

Voip Dis-Advantages:

1) Bandwidth.
2) Electric Power

Voip Protocol Layers:

SIP Protocol Overview

SIP is signalling protocol used to Initiate,Modify and Tear down the

Voice and Video sessions.

SIP is and application layer protocol (also called Signalling/Text-

based protocol). It is designed by IETF. Sip was initially focused on Voice
Communication and later expanded to Video, Instant messaging.....etc. SIP carries
media transport through the RTP/ RTCP, SRTP.

SIP MESSAGES

Sip Messages ae divided into two types:

Sip Requests --> sent from client to sever.

Sip Responses -->sent from server to client.

Message Syntax:

Start line
Message header
CRLF- Carriage return line feed.
Message body(SDP-Session Describestion Protocol ).

Message header: It provides additional information, regarding requests and

responses.

CRLF : Empty line after header field.

Message body( SDP) : Normally describes the type of session to be

established,including a description of media to be exchanged.

Start Line:
Start line can be request line or status line. Sip Requests start line:
Syntax: Method Sp Request URISp SIP version
Ex: Invite sip:bob@billoxi.com sip/2.0

Sip Response start line:

 Syntax:Sip version Sp Response code Sp Responpharse
Ex: sip/2.0 200 OK

Message header Syntax:

Via
To
From
Call Id
C-seq
Max-forward
Contact

Message Body Syntax:

V- Version SDP = 0
O- Owner = IPv4
S- Session name = VOIP
T- Time= Start and stop time
M- Media =Audio/ Video
A- Attributes =Send only/Recv only

REQUESTS:

Invite: It indicates client is invited to participate in call session.

Ack: It confirms client has received final response to invite request.
Bye: Terminates the call and can be sent by either caller or callee.
Cancel: Cancels any pending request.
Options: Queries the capabilities of server.
Register: Registers the address listed in to header field with a Sip server.
Prack: Provisional Acknowledgement.
Refer: Asks recipient to issue SIP request.
Notify: Notify the subscriber of a new event.
Subscribe:Subscribes for an event of notification from the notifier.
Info : Sends mid-session information.
Message:Transports instant messages using SIP.
Update: Modifies the state of a session without changing.
Publish: Publish an event to the server.

RESPONSES:

1XX - Information Responses

2XX - Successful Responses
3XX - Redirection Responses
4XX - Client failure Responses
5XX - Server failure Responses
6XX - Global failure Responses

Examples:

100 - Trying
180 - Ringing
181 - Call being forwarded
183 - Session in progress
200 - Ok
 202 - Accepted
301 - Moved permanently
302 - Moved temporarily
401 - Unauthorized
404 - Not found
483 - Too Many Hops
486 - Busy here
407 - Proxy authentication required
408 - Request timeout
487 - Request terminated
491 - Request pending
503 - Service unavailable

EXPLANATION OF MESSAGE HEADER Fields:

VIA: It contains the address at which user is expecting to receive response to this
request.
The Via header field value MUST contain a branch parameter.

This parameter is used to identify the transaction created by that request.

It helps proxies to detect loops.

The branch parameter value MUST be unique across space and time for all requests
sent by the UA. The exceptions to this rule are CANCEL and ACK for non-2xx
responses. As discussed below, a CANCEL request will have the same value of the
branch parameter as the request it cancels.
* The branch ID inserted by an element always begin with the characters "z9hG4bK".
These 7 characters are used as a "Magic cookie"

FROM :It contains display name and SIP URI that indicates the originator of the
request. The From field MUST contain a new "tag" parameter, chosen by the UAC.

TO : It contains the display name and SIP URI towards which request was originally
directed. A request outside of a dialog MUST NOT contain a To tag; the tag in the
To field of a request identifies the peer of the dialog. Since no dialog is
established, no tag is present.

TAG : Tag in 'To' header are of no help since they are not known until response
arrive.
An initial request from a client will contain a From Tag and the subsequent
provisional response to it from the server will contain a To Tag.

Tags are used by the UAC to distinguish multiple final responses from different
UAS.

CALL-ID: It contains a globally unique identifier for all requests and responses
sent by either UA in a dialog,generated by the combination of random string and
IPaddress.Note that when request are retried after certain failure, These retried
requests are not considered new requests,and therefore do not need new Call-ID.

* Implementations MAY use the form "loclid@host".

* It provides some protection against session hijacking and reduces.

CALL LEG : The combination of to tag, From tag and call ID is called call leg/peer
to peer connection.
C- SEQ :It contains an sequence number and method Name. The c-seq is incremented
for each new request.
* It is used to identify and order transactions.

CONTACT :It contains a SIP URI that represents the direct route to contact user.

MAX-FORWARD:It serves to limit the number of hops a request can make on the way to
destination. If the Max-Forwards value reaches 0 before the request reaches its
destination, it will be rejected with a 483(Too Many Hops) errors response.

CONTENT LENGTH :It contains an octet(byte) count of the message body.

CALLER ID:It is provided by the From SIP header containing the caller's name and
number.

RECORD ROUTE: Record route header is inserted into requests by proxies that want to
be in the path of subsequent request for the same call-id. It is then used by user
agent to route subsequent requests.

SIP ENTITIES:

SIP network consists of four types of

logical SIP entities. Sip entity as a client (initiates requests), and as a
server (responds to requests), or as both.

Following are the four types of logical SIP entities:

USER AGENT : User Agent (UA) is the endpoint logical entity. User Agents initiate
and terminate sessions by exchanging requests and responses.

User Agent Client (UAC)�

The client application that initiates SIP requests(Initiates a call).

It lasts only for the duration of that paticular transaction.

User Agent Server (UAS)�

It generates a response to a SIP request send by UAC.

It lasts only for the duration of that paticular transaction.

Note: The User Agent initiating a call acts as a UAC when sending the initial SIP
request (INVITE) and as a UAS when it receives a SIP BYE request from the callee.

Some of the devices that can have a UA function in a SIP network are: IP-phones,
telephony gateways, call agents, automated answering services.

REGISTRAR SERVER:
Before endpoints communicate (endpoints are
the caller and the callee i.e UA) they should be registered to a SIP entity known
as REGISTRAR SERVER. It is a UAS and a logical entity.The endpoint registers to a
REGISTRAR Sever and Contact Information of the user specified in the request is
updated at Location Server.

A Location Server is a database of locations of SIP User Agents.

1) It is updated by SIP User Agents by Registration.
2) It is used by Redirect/Proxy server to obtain information about a callee's
possible locations.
3) DNS query is used to query location service.

Note: In this both To & From address will be same.

REDIRECT SERVER: Redirect Server is a server that accepts a SIP request and send
3XX Responses with present location address after quieries with Location sever .
Unlike Proxy servers, Redirect Servers do not pass the request on to other servers.

Ex: 302 - Moved Temporarily,301 - Moved Permanently

PROXY SERVER:
A Proxy Server is an intermediary entity that acts as both a
server and a client for the purpose of making requests on behalf of other clients.
It also consults database such as DNS and Location Server.

There are two types of poxies:

Normal Proxy Sever(Sateless):
It just forwards the received requests to other end and send
responses on behalf of other.It just perform routing logic,send message out.

B2BUA Sever(Statefull):
Proxy interprets, and, if necessary, rewrites a request
message before forwarding it. It maintain state during entire transaction.
Ex: Forward on no reply,Forking.

DNS Server: It stores address and its corresponding name pairs. If we send a
website name in a request and it returns exact IP address of it.

SIP Trapezoid:

Basic SIP Trapezoid

biloxi.com
atlanta.com

User A initiates an INVITE request with Request-URI of B(atlanta.com). As invite

reaches "biloxi.xom" proxy server it locates the poxy server at "atlanta.com"
possibly by performing a particular type of DNS lookup and find the server that
serves UserB. Since proxy is providing "outbound service" it is called Outbound
Poxy.
After Invite reaches atlanta.com proxy server consults
database,generically called a location service,that contains the current IP address
of UserB.Since proxy is providing "Inbound service" it is called Inbound Poxy.

A single proxy can have the logic to act as an Outbound/Inbound proxy for a A to B
call.

Difference between H.323&SIP

1. H.323 :It is designed by ITU

SIP :It is designed by IETF.
2. H.323 : H.323 is limited to conferencing.It define the basic set of
funtionality that all devices must support.
SIP : SIP was initially focused ib voice connection and then expanded to
video,instant messaging..etc

3. H.323 : Has defined a number of features to handle failures of N/W entities

SIP : Has not defined for handling device failure. user agent has to send
re-Invite.

4. H.323 : Encodes messages in a compact binary format.

SIP : Sip messages are encoded in ASCIT text format.

5. H.323 : Media transport RTP/RTCP,SRTP.

SIP : Media transport RTP/RTCP, SRTP.

6. H.323 : Addressing H.323 support these aliases:

* E.164 dialed digits
* Transport address
* Email address
* Party number
SIP : SIP only understand URI- Style addresses.

7. H.323 : Call setup : Setup Connect Ack

SIP : Call setup : Invite 200Ok Ack

8. H.323 : H.323 fully support video and data conferencing.

SIP : SIP has limited support for video and no support for data
conferencing.

9. H.323 : H.323 Support any codec standedized or propretory.

SIP : SIP supports IANA registered codec.

10. H.323 : Most H.323 entities use a reliable transport for signalling.
SIP : Most SIP entities uses an unreliable transport for signalling.

11. H.323 : Routing gatekeepers can detect loops by looking at call identifier
and destination address.
SIP : The via header facilitates detecting loops.

12 . H.323 : Minimum ports for VOIP call 3 ( call signalling, RTP, RTCP )
SIP : Minimum ports for VOIP call 3 ( SIP, RTP, RTCP )