cmpe 208 new draft lal

8/14/2019 Cmpe 208 New Draft Lal

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Department of Computer Engineering

CmpE208- Network Architecture and Protocols

(Spring 2009, Tuesday Class)

SIP

Instructor

Prof. Dr. Rod Fatoohi


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Abstract

In this project we propose to perform in-depth analysis of the SIP, Session Initiation Protocol which is anapplication layer control protocol. Voice over IP is a very upcoming technology widely used in

performing PC-PC, PC to PSTN and PSTN to PC calls due to usage of low bandwidth and integration ofvoice with data. Most of the VOIP applications use SIP, as its signaling protocol in performing the basicoperations in a call establishment which are creation, management and termination of sessions betweenhosts in any IP based network. A detailed study of the SIP functionality and its structure is performedfollowed by the experimental analysis of the SIP protocol which is done by using the two most widelyused VOIP applications, Yahoo Messenger and Media Talk Ring. Test connections are set up between twohosts using the above mentioned applications and the data packets are captured using the Wireshark toolwhere the various SIP methods and messages are presented. Some of the test scenarios such as call setup,call forwarding on no answer, call placed with a request URI where the user does not exist, user located atanother location, called party is busy, called party terminates the call are performed by setting up callsbetween PC-PC and PC-PSTN users. SIP works in conjunction with another protocol called SDP, theSession Description Protocol which is used to share the session information between the two hosts. Adetailed study of the SDP is also performed with respect to the RFCs and a detailed analysis is done and

verified.

IntroductionSIP is a TCP/IP based Application Layer control protocol which initiates, manages and terminatesmultimedia sessions. SIP runs on TCP, UDP and is designed to be independent of the underlying transportlayer. It is a text based Protocol similar to the Hypertext Transfer Protocol. Participants can be invited toalready existing sessions using SIP. It developed by the SIP working group within the Internet engineeringTask Force (IETF) multimedia data and control architecture. It is used in conjunction with SessionDescription Protocol (SDP) which is an IETF protocol.

SIP Architecture

SIP consists of two basic components: the SIP clients and SIP servers. The SIP clients are also knows asUser-Agent clients (UAC) is an end system component to initiate a call or send SIP requests. The SIPserver also known as User Agent Server (UAS) is a network device that answers the call or responds tothe requests. SIP is a client-server Protocol.

SIP network consists of four different servers Proxy server, redirect server, user agent server, and registrarserver.

User Agent:

It is an endpoint entity. It starts and ends the session by exchanging requests and response.It is a client who is starting a session between serverand client. This client is always a sender in sip server while server is a receiver. This client can be phone, fax or anything. According torequirements the role of server and client can be interchanged in the single session.

Registrar:

In SIP registrar has the same functionality as Gate Keeper has in H.323. This register is working as a database center in SIP Server.It has all the information like IP address and port address of all connected users. It is also updating its data base according to theconnection changes between server and client. It is performing an authentication function.

Proxy Server:

Proxy Servers execute call signaling on the behalf of the parties they serve. It is also performing an authentication function. Proxy Server stay inthe path of call routing. It receives SIP requests and routes them to the other users or other Proxy Servers. Server do it based on therouting data base and routing rules. Proxy Server can stay in the path of call.

Redirect Server:

Team Members:

Lalitha Garudadri SJSU Id: 006196034

Varun Kalyan Anantharaman SJSU Id: 005889377 < [email protected]>


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Redirect Server determine the current location of the called party and gave its address to the calling party . Then calling party can dodirect contact of called party. It is basically used to point the location purpose. Redirect server is not staying in the path of call.

SIP Advantages

SIPProtocol StructureThe syntax of the SIP protocol is text based and looks similar to the HTTP. SIP messages are broadlyclassified into two types SIP request messages which are sent by the client to the server, and SIP responsemessages or status messages which are sent by the server in response to the client. Both the SIP requestsand SIP response messages have a similar syntax and consist of

Start line: The start line consists of the request-line and status line. The request line specifies thetype of request and the status line gives the response for the request a success or a failure.

Zero or more headers: This field gives additional information like the message originator, messagerecipient, retry-after header and subject header.

Message body: Consists of SDP which provides format for describing the session informationbetween the session participants.

SIP Request Messages

The SIP request message consists of a request line which includes the following fields a methodtoken, a request URI and SIP version. Request URI is the address of the user to which request is sent,SIP version specifies the version of SIP being used. The method token describes one of the sixdifferent SIP methods that are defined in the RFC. The following captures which are obtained byplacing calls between two VoIP clients using Yahoo Messenger and MediaRing Talk are used toexplain the various SIP request messages that are defined in the RFC.

INVITE: The SIP INVITE method is used to start a session or place a call. This method consists ofthe following fields

Request-line: Which shows the SIP method INVITE, request URI i.e., address of the

called party sip:[email protected]:5050and SIP version transport=tcp sip/2.0 Message Header: The message body consists of From, To, Call-ID, Cseq, User-Agent, Via,

Contact, Allow, Content type, and Content-length fields.

Message Body: It consists of the SDP which provides format for describing the session

information like session name, session originator, media type etc between the sessionparticipants.

Capture: Invite Method
http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:5050http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:5050http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:5050http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:5050


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ACK: This method is sent by the calling party when a final response like called user is busy or callcannot be complete is received in response to the INVITE method. This method contains thefollowing fields Request-line, Message header From, To, Call-ID, Cseq, User-Agent, Max-Forwards, Via, Contact, Content-Length. It can be observed from the capture that the contentlength is zero as the message body is absent.

Capture: ACK Method

CANCEL: This method is used to terminate a session for which an INVITE method is sent but afinal response is not received. The Content-length field is zero as observed from the captures asthere is no message body.

Capture: CANCEL Method

BYE: This method is used to terminate the session. It can be initiated by the calling or the calledparty. The Content-Length filed is set to zero as there is no message body.

Capture: BYE Method

OPTIONS: This method can be used to find the features like type of media supported by the calledparty by enquiring the server.

REGISTER: This method is used by the user agent client to let the register server know where it islocated so that it can be contacted. The UAC registers its address with the SIP server for thispurpose. It can be observed that To and the From fields are the same as the user is registeringhimself. The Register messages for one user should use the same call ID. The content length is setto 0 as there is no message body. The Expires field is set to 3600sec indicating the duration forwhich the registration is valid.

Capture: Register Method


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Additional SIP methods SUBSCRIBE: This method is used by the client to subscribe to certain events. It can be observed

from the capture that the Event field in this case is set to billing.

Capture: SUBSCRIBE method

NOTIFY: When the events occur the client is informed using the NOTIFY method for the events itsubscribed to. It can be observed from the capture that the Event field is set to billing to the

corresponding SUBSCRIBE request.

Capture: NOTIFY Method

INFO: This method is used to communicate the mid-session signaling information by carryingthe session-related control information. This method does not changes the state of SIP calls andthe state of sessions initiated by SIP. It is used to provide more information which can enhancethe applications using SIP.

Capture: INFO Method

SIP Response Messages


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The SIP response message consists of a status line which is a start line. The status line comprises of SIPversion, a three digit number which indicates the result of the request called status code and reason phrasethat interprets the textual description of the result. The RFC defines the following SIP response messages.These responses can be explained from the captures obtained by placing a call between two clients usingYahoo Messenger PC-PC and one client using Yahoo Messenger and the other using PSTN mobile servicePC-PSTN.

Provisional (1xx): Indicates that request sent by the client is received by the server and it isprocessing the request. TRYING 100, RINGING 180, CALL IS BEING FORWARDED 181,QUEUED 182, SESSION PROGRESS 183 are the 1xx responses defined in the RFC

TRYING 100: The TRYING response is sent by the proxy server that serves the called party

in response to the INVITE method sent by the proxy server that serves the calling party. Thisresponse is sent to the calling party to indicate that the INVITE method is received and theproxy is working to route the INVITE message on its behalf. It can be observed that theContent-Length field is zero.

Capture: 100 TRYING

RINGING 180: This response is sent back to the calling party which indicates that the

INVITE method is received and the called party is alerted.

Capture: 180 RINGING

CALL IS BEING FORWARDED 181: This status code is used by the server to indicate that

the call is forwarded to another destination example a voice mail server

Capture: 181 CALL IS BEING FORWARDED

SESSION PROGRESS 183: The 183 Session Progress Message is a new extension of SIP

response which sends an indication from the calling user agent client to the called user agentserver that a One-way media path is opened from the called party to the calling party.

Capture: 183 SESSION PROGRESS

Success (2xx): This response indicates that the server received, accepted and understood theactions successfully. OK 200 is the 2xx response defined by the RFC

OK 200 : This status code indicates that the request is successful


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Capture: OK 200

Redirection (3xx): These responses indicate that further actions are needed to complete therequest.

Request Failure (4xx): These responses indicate that there is a failure from a particular server.The request by the client might contain a bad syntax or cannot be fulfilled at the server. The clientcan modify the request and try again. UNAUTHORIZED 401, FORBIDDIEN 403, PROXYAUTHORIZATION REQUIRED 407, TEMPORARILY UNAVAILABLE 480, REQUESTTERMINATED 487, REQUEST PENDING 491 are few of the responses defined in the RFC

UNAUTHORIZED 401: This response indicates that the request needs user authentication. It

is issued by User agent server and registrars.

Capture: UNAUTHORIZED 401

FORBIDDEN 403: This response is sent by the server indicating that it understood the clients

request but is refusing to fulfill the request.

Capture: FORBIDDEN 403

NOT FOUND 404:

This response is sent back to the user indicating that the server has information that theuser does not exit at the domain that is specified in the Request-URI

Capture: NOT FOUND 404

PROXY Authentication Requires UNAUTHORIZED 407:This response indicates that the

client should first authenticate itself with the server.


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Capture: PROXY Authentication Requires UNAUTHORIZED 407

TEMPORARILY UNAVAILABLE 480: This response indicates that the called party is

contacted successfully but is temporarily not available. For example the user could beavailable in some other location unknown by the server, the user is not logged in, or the useractivated do not disturb feature.

Capture: Temporarily unavailable 480

CALL LEG/TRANSACTION DOES NOT EXIST 481:This response indicates that a request

which does not match any existing transaction is received by User Agent Server.

Capture: Call Leg/ Transaction Does Not Exist 481

ADDRESS INCOMPLETE 484: Indicates that a request is received by the server with an

incomplete Request-URI

Capture: Address Incomplete 484

BUSY HERE 486: Indicates that the called party is busy

Capture: Busy Here 486

REQUEST TERMINATED 487: This response is sent when the request is terminated by a

CANCEL or BYE request.

Capture: REQUEST TERMINATED 487


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REQUEST PENDING 491: This response is sent by the User agent server to the client

indicating that the request was received by the UAS and has a pending request within the samedialog..

Capture: REQUEST PENDING 491

Server Error (5xx): These responses indicate that the server failed.

BAD GATEWAY 502: This response indicates that the invalid response is received by

the server when it is trying to fulfill a request from a downstream server.

Capture: Bad Gateway 502

SERVICE UNAVAILABLE 503: This response indicates that the server cannot respond

temporarily due to maintenance or overloading of the server

Capture: Service Unavailable 503

SERVER TIME-OUT 504: A response is not received on time by the external server in

the attempt of processing the request.

Capture: Server Time-out 504

Global Failure (6xx): These responses indicate that the server has information about the user butthe request cannot be fulfilled at any server.

SIP Scenarios1. Simple Call flow

Call InitiationThis scenario is used to show the basic function of exchange of SIP messages used to initiate asession. To study the exchange of messages a call is setup between two clients using SIP YahooMessenger VoIP application. A call is placed between the two clients using SIP identity i.e..,


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SIP URI which is Uniform Resource Identifier. The exchange of the SIP messages between thetwo clients is shown in the figure below.

Capture: Showing the exchange of SIP messages for establishing a call

Timeline Capture: Showing the exchange of SIP messages for establishing a call

The SIP URI is similar to an Email address which consists of a username and host name orUsername and Phone. It can be observed from the capture that SIP URI in this scenario issip:[email protected]:443 where 68.142.233.115:443 is the domain of thecalled party SIP service provider.

INVITE MESSAGE: SIP is a request/response transaction model. The transaction is initiated

when the calling party sends an INVITE request to the called party using the called parties SIPURI. The INVITE request contains a number of fields. The fields are described below.

Capture: Invite messageThe first line shows the Request line which contains the INVITE method. Following theRequest line is the Message header indicating a list of header fields which are
http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443


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Via: TheVia field contains the sip version being used, the address at which the calling

party wishes to receive the response for the request 192.168.1.107:55154. It also containsa branch parameter indentifying a transaction.

Contact: This field contains the SIP URI that indicates a direct route to contact the calling

party. Contact header is used to let others know where to send future requests.

To: This field provides the address of the called party i.e., the called users namegarudadrikrishnamurthy and SIP URI sip:[email protected]:443

From: This field provides the address of the calling party i.e.., the calling users name

glalitha21 and SIP URI sip:[email protected]:443who initiated the sessionfollowed by a tag field 207b1657 which is generated by the calling party for identificationand is a pseudorandom value.

Call-ID: A unique ID set by the calling party which uniquely identifies an invitation to a

session. Cseq: This field specifies the method being used and has a unique integer number the Cseq

integer value will be incremented for same Call-ID initiating a new request method. User-Agent: This field indicates the type of application being used.

Content type: This field indicates the media type of message body. Most of the VoIP

applications have this field as SDP. Content-length: This field indicates the length of the message body in octets

It can be observed from the To and From field that the proxy server for both the callingand called party is the same.

TYRING message: When the INVITE message is received by the proxy server

(68.142.233.155) it acts on behalf of the called party and sends a TYRING(100) response backto the calling party.

Capture: Trying messageThis response indicates that the INVITE message has been received and the proxy serveris working to route that message to the destination. It can be observed from the INVITEcapture that From, To, Call-ID, Cseq, Branch field in the Via filed is the same as theINVITE message to help the calling party to correlate the response to the INVITE

message sent.

RINGING message: When the INVITE message is received by the called party the called

party is alerted to the incoming call and if the called party decides to receive the call a 180ringing response is sent back to the calling party.
http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443http://var/www/apps/scribd/scribd/tmp/scratch3/sip:[email protected]:443


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Capture: Ringing Message

OK message: When the called party receives the call 200 OK response is sent by the called

party indicating that the call is answered. This response contains a message body with the SDPmedia description which indicates the type of session the called party is ready to establish withthe calling party.

Capture: OK message

ACK message: An Ack message is sent by the calling party to the called party to confirm the

reception of the final response.

Capture: Ack Message

Session Description ProtocolIt can be observed that there is a two-phase exchange of the SDP messages between the two parties onefrom the calling to the called party and another from the called to the calling party providing thenegotiation capabilities between the two end parties. SDP information is carried in the SIP message body.The session information between two participants is described using the Session Description Protocol.The parameters relating to the media stream is described using session-level parameters and media-levelparameters. The session-level parameters include the following information protocol version(v), name ofthe session(s), originator session ID(o), time of the session(t) i.e. how long the session should be active.


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The media-level parameters include the following information like media description which comprises ofthe media type, port number, transport protocol, media format (m), media information(i), connectioninformation(c), bandwidth information(b), attributes(a). These fields can be observed in the followingSDP capture.The syntax of all the fields are described below

Origin (o): Consists of 6 subfields username, session ID, version, network type (IN), address

type (IP4) and address of session originator. Connection Data (c): Consists of three subfields network type (IN), address type (IP4),

connection address of the session originator.

Media Information (m): Consists of four subfields media type (audio), port (45825), transport

(RTP/AVP), formats lists all various type of media format that a user can support.

Attributes (a): Can be specified at the session level or media level or both. Attributesare of two forms property attributes and value attributes. Property attributes specifiesthat a session has a specific characteristic like sendonly or recvonly. Value attributeslike orient specify that a session has a particular characteristic of a particular value. Format ofattribute is a: rtpmap:

Example a: rtpmap: 103 ISAC/16000

Capture: Session Description Message send from calling party to called party


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Capture: Session Description Message send from called party to calling party

Call termination:The capture below shows the exchange of SIP messages when a party wishes to terminate a call.

Capture: Showing exchange of SIP messages to terminate the call

Timeline capture: Showing exchange of SIP messages to terminate the call

BYE message: One of the parties generates a BYE message when it is decided to terminate the

call. In this scenario the calling party generates a BYE message.


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Capture: BYE message

OK message: The called party sends a 200 OK message to confirm receipt of the BYE

message which terminates the session.

Capture: OK message

2. Call hold

In this scenario a call is placed is placed to one client using Yahoo Messenger and the call isplaced on hold and another call is made to second client. The second call is terminated and thefirst call is resumed back. Finally the first call is terminated. The Exchange of SIP packets isshown in the capture below.


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3. Call Forwarding on No Answer: - 181 response

This capture is obtained when a call is placed between two Yahoo Messenger SIP clients and the calledparty does not receive the call. The call is being forwarded to an automatic voice message system.


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Capture: Showing the exchange of SIP messages

Capture: Call Is Being Forwarded 181

The timeline graph below shows the exchange of SIP messages for the call set up. The call establishmentprocess is the same as explained in the first scenario. When a ringing 180 message is sent from the calledto the calling party it is up to the called party to answer the call or ignore the call. When the called partydoes not receive the call a 181 message is sent from the called to the calling party.

Capture: Showing the exchange of SIP messages when the called party does notreceive the call (181 call is being forwarded message).


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4. Call placed to an Invalid number

The following capture is obtained when a call is placed to an invalid number a Call Forbidden 403response is sent back to the calling party indicating that its an invalid number.

Capture: Forbidden 403 Invalid number

The capture below is obtained by placing a call from Yahoo Messenger which does not have a phone

out service. When the user tires to place the call a Forbidden 403 message is also sent by the serverindicating that Subscription is required.

Capture: Forbidden 403 Subscription Required

5. Call placed with a Request-URI where the user does not exist.

The following capture is obtained by placing a call using MediaRing talk to a number which does notmatch with any of the domains that the recipient of the request handles. A Not Found 404 response issent back to the calling party indicating that the user does not exist at the domain specified in theRequest URI.

Capture: Not Found 404


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Capture: Showing the exchange of SIP messages when user does not exist at the domain specifiedin the Request URI (404 Not Found).

6. User is located at another location - Temporarily Unavailable 480:This response is obtained when a call was placed by using the Yahoo Messenger phone out service. Aphone number was registered twice using two different names. The response is forwarded by theserver to the calling party indicating that the Called party is contacted successfully but the calledparty is currently unavailable as the user is available at another location.

Capture: Temporarily Unavailable 480

Capture: Showing the exchange of SIP messages when the called party is available at anotherlocation (480 Temporarily Unavailable).

7. Call Placed with Request URI that does not match transaction

When a call is placed from Yahoo Messenger with an ID that does not have the phone out service with anumber that doesnot match any transaction or match any existing numbers the server sends back CallLeg/Transaction does not exist to the calling party. The response can be observed from the capture below.


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Capture: Call Leg/ Transaction Does not Exist

Capture: Showing the exchange of SIP messages when the number does not match any transaction(481 Call Leg/ Transaction Does not Exist).

8. Address Incomplete: Call placed with Request-URI incomplete:The capture below is obtained by placing a call from Yahoo Messenger by using the phoneout service to a number with incomplete information like dialing few digits the server sendsback an Address Incomplete 484 response indicating that the server received a request with aRequest-URI which is incomplete. Additional information needs to be provided to completethe call.

Capture: Address Incomplete 484


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Capture: Showing the exchange of SIP messages Call placed with incomplete URI (484 AddressIncomplete)

9. Call Party is BusyThe following capture is obtained when a call is placed to a client which is currently not willing toaccept the call or cannot accept any additional call as it is on a call a Busy Here 486 response is sent

back by the server to the client indicating that the called party is busy.

Capture: Busy Here 486

Capture: Showing the exchange of SIP messages when the called party busy (486 BusyHere).

10. Called party terminates the call

When a call is placed between two clients the called party can receive the call or reject thecall. The following capture is obtained when a call is placed between two Yahoo Messengerclients and the called party rejects the call. A status code 487 request terminated message issent by the server to the calling party indicating that the called party rejected the call.

Capture: Request Terminated 487


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Capture: Showing the exchange of SIP messages when the called party terminates the call (487Request Terminated).

11. 502 Bad GatewayThefollowing capture is obtained by placing a call using MediaRing Talkwhen the serverreceives an invalid response from a downstream server in the process of fulfilling the requestby the calling party a Bad Gateway 502 response is sent back to the calling party.

Capture: Bad Gateway 502

Capture: Showing the exchange of SIP messages when server receives invalid responsefrom downstream server (502 Bad Gateway).

12. Service Unavailable

When a call is placed to a mobile from a VoIP application like SKYPE or Yahoo Messenger a defaultnumber is displayed for that call. The following capture is obtained when a call is placed from MediaRing talk to that default number. When such a call is placed by the calling party the server respondedwith a 503 service unavailable message indicating that the server is temporarily unable to process therequest due to maintenance or temporary overloading of the server.


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Capture: Service Unavailable 503

Capture: Showing the exchange of SIP messages when the server is Temporarily Unavailable (503Service Unavailable).

13. Server Time-out:

When the server does not receive a timely response when it tries to process the request from anexternal server a Server Time-out 504 response is sent back to the calling party as shown in thecapture below.

Capture: Server Time-Out 504

Capture: Showing the exchange of SIP messages when the Server does not receive a timely response(504 Server Time-Out).

14.SIP 183 Session Progress Message - PSTN Internetworking


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The 183 Session Progress Message is a new extension of SIP response which sends an indicationfrom the calling user agent client to the called user agent server that a One-way media path is openedfrom the called party to the calling party. In-band call progress information such as tones andannouncements are provided to the calling client using this process. In traditional telephone networkthis requirement is handled with SS7 using the Address Complete Message (ACM).

PSTN internetworking concept is described using a test scenario when a call is placed from Yahoomessage (SIP client) to a mobile phone (PSTN Network). The SIP network should be able tocommunicate with the PSTN when a call is established between a SIP-based network and PSTNusing the signaling protocol used by the PSTN which is the SS7 protocol. For internetworkingbetween the SIP and the PSTN there is a Gateway entity in the network to provide the conversionfrom circuit-switched media to packet-switched media.

When a call is placed from Phone out service provided by the Yahoo messenger to Mobile phoneconnected to the PSTN switch the exchange of SIP messages takes place as shown in the capturesbelow. It can be observed that the SIP URL has the form of a telephone number which can be viewedin the To field in the INVITE message. The INVITE message sent by the calling party to establish asession is mapped to the IAM message of the calling party. The PSTN switch sends an ACM messagein response to the IAM which is mapped to the 183 (session progress) response. The in-bandinformation which is contained in the SDP of the 183 response message is returned from the calledswitch to the SIP caller. This enables a one way path from the Called to the Calling party. An ANM isreturned back which is mapped to the 200 (Ok) responses when the call is answered. An updatedSession description is sent in this response. An ACK message is sent from the calling to the calledparty. Two way voice sessions is established. Any party can terminate the session by sending a Byemessage.

Capture: Showing the exchange of SIP messages.


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Capture: Timeline graph showing the exchange of SIP messages

Capture: Invite message showing the SIP URL has the form of telephone number in the To field

Capture: 183 Session Progress Message

At the end of the session a Notify message is sent back to the caller. The SDP in this messagecontains the billing information: the cost if the call, the remaining balance etc which is shown in thecapture below.


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Capture: Notify message showing the billing information at the end of the session

cmpe 208 new draft lal

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