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Enterprise Communication

SIP – Session Initiation Protocol

Protocol Introduction1

Example Analysis3

Protocol Message2

Tool Use4

OSI TCP/IP Corresponding Protocol

Application

Application

TFTP FTPSNMP SMTP DNS Telnet

SIPPresentation

Session

Transport Transport TCP UDP

Network Internet IP ICMP ARP

Data Link Network HDLC PPP SLIP

Physical

TCP/IP Protocol Stack

SIP – Protocol introduction

1. SIP （ Session Initiation Protocol) is the application layer control protocol for multimedia communications on IP networks, it is used to create, modify, and terminate a session process which has one or more participants.

2. SIP protocol can be used to initiate a session, also used to invite members to join the session which has been established in other ways.

3. Adopting transaction mechanism and operation method of each request triggers the Server. A request and a response constitute a transaction, the transactions are independent of each other.

4. Using textual encoding and decoding.

5. It’s based on the SMTP and HTTP protocols.

6. SIP protocol is only responsible for setting up the call (Signalling) only.

SIP – Correlated protocol

SIP does not describe and transfer the multimedia information. To complete the call, you also need the help of two protocols:

1. SDP （ Session Description Protocol ） is responsible for setting the media attribute.

For example, setting the session port (port) and encoding (codec)

2. RTP （ Real-time Transport Protocol ） is the voice in a call or the video in a video call.

SRTP （ Secure RTP) Since RTP can be intercepted and decoded easily by any network capture. SRTP uses AES as encryption to protect the voice.

RFC(Request for Comments) LUT

SIP - RFC 3261 (There are many extended SIP RFC, and the functions are increasing now. ）

SDP - RFC 4566

RTP - RFC 3550

SRTP - RFC 3711

SIP – Correlated protocol

Protocol stack

H.323 SIP RTSP RSVP RTCPH.263 etc.

RTP

TCP UDP

IP

PPP

Sonet

AAL3/4 AAL5

ATM Ethernet

PPP

V.34

SIP – Session Initiation Protocol

Protocol Introduction1

Example Analysis3

Protocol Message2

Tool Use4

User-Agent UA(User Agent) represents a terminal system. It consists of two parts:User Agent ClientA user agent client is a logical entity that creates a new request, and then uses the client transaction state machinery to send it. The role of UAC lasts only for the duration of that transaction. In other words, if a piece of software initiates a request, it acts as a UAC for the duration of that transaction. If it receives a request later, it assumes the role of a user agent server for the processing of that transaction.

User Agent ServerA user agent server is a logical entity that generates a response to a SIP request. The response accepts, rejects, or redirects the request. This role lasts only for the duration of that transaction. In other words, if a piece of software responds to a request, it acts as a UAS for the duration of that transaction. If it generates a request later, it assumes the role of a user agent client for the processing of that transaction.

Transaction and dialog TransactionSIP is a transactional protocol: interactions between components takeplace in a series of independent message exchanges. Specifically, aSIP transaction consists of a single request and any responses tothat request, which include zero or more provisional responses andone or more final responses. In the case of a transaction where therequest was an INVITE (known as an INVITE transaction), thetransaction also includes the ACK only if the final response was nota 2xx response. If the response was a 2xx, the ACK is not consideredpart of the transaction. We can use branch to identify different transaction.

DialogDialog: A dialog is a peer-to-peer SIP relationship between two UAs that persists for some time. A dialog is established by SIP messages, such as a 2xx response to an INVITE request. A dialog is identified by a call identifier, local tag, and a remote tag. A dialog was formerly known as a call leg in RFC2543. We can use Call-ID to indentify different dialog.

Transaction and dialog

SIP Message

SIP is a text-based protocol and uses the UTF-8 char set. A SIP message is either a request from a client to a server, or a response from a server to a client.

Request: A SIP message sent from a client to a server, for the purpose of invoking a particular operation.

Response: A SIP message sent from a server to a client, for indicating the status of a request sent from the client to the server.

SIP - Request Message SIP requests are distinguished by having a Request-Line for a start

line. A Request-Line contains a method name, a Request-URI, and the protocol version separated by a single space (SP) character.

The Request-Line ends with CRLF. No CR or LF are allowed except in the end-of-line CRLF sequence. No linear whitespace (LWS) is allowed in any of the elements.

Request-Line = Method SP Request-URI SP SIP-Version CRLF

Method: This specification defines six methods: REGISTER for registering contact information, INVITE, ACK, and CANCEL forsetting up sessions, BYE for terminating sessions, and OPTIONS for querying servers about their capabilities. SIP extensions, documented in standards track RFCs, may define additional methods.

SIP - Request MessageRequest-URI: The Request-URI is a SIP or SIPS URI or a general URI (RFC 2396 [5]). It indicates the user or service to which this request is being addressed. The Request-URI MUST NOT contain unescaped spaces or control characters and MUST NOT be enclosed in "<>“. SIP elements MAY support Request-URIs with schemes other than "sip" and "sips", for example the "tel" URI scheme of RFC2806 [9]. SIP elements MAY translate non-SIP URIs using any mechanism at their disposal, resulting in SIP URI, SIPS URI, or some other scheme.

SIP-Version: Both request and response messages include the version of SIP in use, and follow [H3.1] (with HTTP replaced by SIP, and HTTP/1.1 replaced by SIP/2.0) regarding version ordering, compliance requirements, and upgrading of version numbers. To be compliant with this specification, applications sending SIP messages MUST include a SIP-Version of "SIP/2.0". The SIP-Version string is case-insensitive, but implementations MUST send upper-case. Unlike HTTP/1.1, SIP treats the version number as a literal string. In practice, this should make no difference.

SIP - Request MessageRequest message The meaning of the message

INVITE

Initiate a session request, invite users to join a session, the session description is contained in the message body. For two-party call, the caller indicated the acceptable media type and parameters in the session description. The callee must show the media type which does it want to accept in the message body of the successful response and also instruct the media which will be send. If you receive an invitation to attend the meeting, the callee can determine the user has joined the conference according to the Call-ID or the identity of the session description , and return the success response message.

ACKConfirmed receive the final response for INVITE request. The message only can be used with INVITE message.

BYE End the session.

CANCELCancel the uncompleted request, the completed request (ie receive a final response for the request) is not affected

REGISTER Register

OPTIONS query the server ability

SIP - Request Message Format

Command name Request URI Protocol

version

Call-id: value

Via: value

From: value

To: value

Contact: value

Cseq: value

Content-Length: value

Max-Forward: value

Content-Type: value

CRLF

SDP

Parameter line

Headers

SIP – Request Message Format

REGISTER sip:192.168.5.139 SIP/2.0Via: SIP/2.0/UDP 192.168.4.16:5029;branch=z9hG4bK273563898From: "502" <sip:502@192.168.5.139>;tag=1411642056To: "502" <sip:502@192.168.5.139>Call-ID: 143870753@192.168.4.16CSeq: 2301 REGISTERContact: <sip:502@192.168.4.16:5029>Authorization: Digest username="502", realm="asterisk", nonce="3a156c7a", uri="sip:192.168.5.139", response="1ffdb7809cda0624100a0524af26944c", algorithm=MD5Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGEMax-Forwards: 70User-Agent: Yealink SIP-T28P 2.50.0.50Proxy-Require: com.nortelnetworks.firewallExpires: 300Mac: 001565111886Line: 4Content-Length: 0

SIP - Response Message SIP responses are distinguished from requests by having a Status-Line as their start-line. A Status-Line consists of the protocol version followed by a numeric Status-Code and its associated textual phrase, with each element separated by a single SP character. No CR or LF is allowed except in the final CRLF sequence.

Status-Line = SIP-Version SP Status-Code SP Reason-Phrase CRLF

The Status-Code is a 3-digit integer result code that indicates the outcome of an attempt to understand and satisfy a request. The Reason-Phrase is intended to give a short textual description of the Status-Code. The Status-Code is intended for use by automata, whereas the Reason-Phrase is intended for the human user. A client is not required to examine or display the Reason-Phrase. While this specification suggests specific wording for the reason phrase, implementations MAY choose other text, for example, in the language indicated in the Accept-Language header field of the request.

The first digit of the Status-Code defines the class of response.The last two digits do not have any categorization role. For thisreason, any response with a status code between 100 and 199 isreferred to as a "1xx response", any response with a status codebetween 200 and 299 as a "2xx response", and so on. SIP/2.0 allows six values for the first digit:

SIP - Response Message

SIP – Response Message

Serial number Status code Message capability

1xx Provisional request received, continuing to process the request

2xx Success the action was successfully received, understood, and accepted

3xx Redirection further action needs to be taken in order to complete the request

4xx Client Error the request contains bad syntax or cannot be fulfilled at this server

5xx Server Error the server failed to fulfill an apparently valid request

6xx Global Failure the request cannot be fulfilled at anyserver

SIP/Protocol version Response header

Call-id: value

Via: value

From: value

To: value

Contact: value

Cseq: value

Content-Length: value

Max-Forward: value

Content-Type: value

CRLF

SDP

SIP - Response Message Format

Parameter line

Headers

SIP – Response Message Format

SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 192.168.4.16:5029;branch=z9hG4bK273563898;received=192.168.4.16From: "502" <sip:502@192.168.5.139>;tag=1411642056To: "502" <sip:502@192.168.5.139>;tag=as76ac4feaCall-ID: 143870753@192.168.4.16CSeq: 2301 REGISTERServer: Asterisk PBX SVN--r1972MAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7ce05bb3"Content-Length: 0

1. The Via header field indicates the transport used for the transaction and identifies the location where the response is to be sent. A Via header field value is added only after the transport that will be used to reach the next hop has been selected

2. The From header field indicates the logical identity of the initiatorof the request, possibly the user’s address-of-record. Like the Toheader field, it contains a URI and optionally a display name. It isused by SIP elements to determine which processing rules to apply toa request (for example, automatic call rejection). As such, it isvery important that the From URI not contain IP addresses or the FQDNof the host on which the UA is running, since these are not logicalnames.

SIP – Header Fields Details

3. The To header field first and foremost specifies the desired "logical" recipient of the request, or the address-of-record of the user or resource that is the target of this request. This may or may not be the ultimate recipient of the request. The To header field MAY contain a SIP or SIPS URI, but it may also make use of other URI schemes (the tel URL (RFC 2806 [9]), for example) when appropriate. All SIP implementations MUST support the SIP URI scheme. Any implementation that supports TLS MUST support the SIPS URI scheme. The To header field allows for a display name.

4. The Call-ID header field acts as a unique identifier to grouptogether a series of messages. It MUST be the same for all requests and responses sent by either UA in a dialog. It SHOULD be the same in each registration from a UA.

SIP – Header Fields Details

5. The CSeq header field serves as a way to identify and ordertransactions. It consists of a sequence number and a method. Themethod MUST match that of the request. For non-REGISTER requests outside of a dialog, the sequence number value is arbitrary. The sequence number value MUST be expressible as a 32-bit unsigned integer and MUST be less than 2**31. As long as it follows the above guidelines, a client may use any mechanism it would like to select CSeq header field values.

6. The Max-Forwards header field serves to limit the number of hops a request can transit on the way to its destination. It consists of aninteger that is decremented by one at each hop. If the Max-Forwards value reaches 0 before the request reaches its destination, it will be rejected with a 483(Too Many Hops) error response.

SIP – Header Fields Details

SIP – Common Expended Protocols

As SIP is a protocol in the continuous development, there are many of the expanded protocols. If need to know, please read related RFC documents for more detail:

http://www.voip-info.org/wiki/view/SIP

SIP – Session Initiation Protocol

Protocol Introduction1

Example Analysis3

Protocol Message2

Tool Use4

the example package

SIP Phone

Register

401 Unauthorized

Register

200 OK

MyPBX

SIP – Register Flow

SIP – Call Flow

MyPBXSIP PhoneA SIP PhoneB

INVITE5

100 Trying6INVITE with sdp17

100 Trying8

180 Ringing9180 Ringing10

200 OK sdp212200 OK sdp211

INVITE with sdp11

100 Trying2

4073ACK4

the example package

ACK ACK

SIP – Others

1. Options ： query the server capabilities

2. Subscribe ： subscribe UA status

3. Notify ： notify UA status

Please look at the example package

SIP – Session Initiation Protocol

Protocol Introduction1

Example Analysis3

Protocol Message2

Tool Use4

MyPBX CLI – SIP debug

SIP debug

Tool - Wireshark

Real-time view sip package

SIP Conversation Analysis

SIP – Conversation Analysis