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Technical Manual – Signaling Protocols Contents

Contents

14 SIP .............................................................................................................................................14-1
14.1 Description of SIP .....................................................................................................................................14-2
14.1.1 Related Terms ..................................................................................................................................14-2
14.1.2 SIP Addressing.................................................................................................................................14-4
14.2 SIP Message Types....................................................................................................................................14-4
14.2.1 Request Messages ............................................................................................................................14-5
14.2.2 Response Messages..........................................................................................................................14-5
14.3 SIP Message Structure...............................................................................................................................14-6
14.3.1 Request Message Structure ..............................................................................................................14-6
14.3.2 Response Message Structure..........................................................................................................14-12
14.4 Introduction to SIP-T ..............................................................................................................................14-13
14.5 SIP Signaling Procedures ........................................................................................................................14-14
14.5.1 Flows of Mobile Originated Calls Through SIP Trunks.................................................................14-14
14.5.2 SIP-T Signaling Procedure.............................................................................................................14-16

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Figures

Figure 14-1 Structure of a SIP request message ...............................................................................................14-7

Figure 14-2 Structure of a SIP response message...........................................................................................14-13
Figure 14-3 SIP flowchart of an MOC through SIP trunks ............................................................................14-15

Figure 14-4 A successful SIP-T procedure (PSTN-IP-PSTN)........................................................................14-16

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Technical Manual – Signaling Protocols Tables

Tables

Table 14-1 Request messages ...........................................................................................................................14-5

Table 14-2 Response messages.........................................................................................................................14-6

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14 SIP

About This Chapter

The following table lists the contents of this chapter.

Section Describes

14.1 Description of SIP Application and related terms of SIP protocol.

14.2 SIP Message Types SIP message types.
14.3 SIP Message Structure SIP message structure.
14.4 Introduction to SIP-T Application of SIP-T.
14.5 SIP Signaling Procedures Examples about the SIP signaling procedures.

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14.1 Description of SIP

SIP is an application-layer control protocol that can establish, modify, and terminate sessions
or calls. These sessions include multimedia conferences, Internet telephony, and similar
applications. SIP is one of the key protocols that implement the voice over IP (VoIP).
SIP supports such services as:
z Name mapping
z Redirection
z Integrated services digital network (ISDN) services
z Intelligent network services
These facilities also enable personal mobility. That is, end users can originate and receive
calls and access subscribed telecommunication services in any location at any time.
SIP supports five facets of establishing and terminating multimedia communications:
z User location: determining the end system to be used for communication
z User capabilities: determining the media and media parameters to be used
z User availability: determining the willingness of the called party to engage in
communications
z Call setup: sending ring back tones to the called party and establishing call parameters at
both called and calling parties
z Call handling and control: including redirection, transfer, and termination of calls
SIP can also initiate multi-party calls using a multipoint control unit (MCU) or fully meshed
interconnection instead of multicast. Internet telephony gateways that connect public switched
telephone network (PSTN) parties can also use SIP to set up calls between them.
SIP makes minimal assumptions about the underlying transport and network-layer protocols.
The lower layer can provide either a packet or a byte stream service, with reliable or
unreliable service. SIP can use user datagram protocol (UDP) and transmission control
protocol (TCP) as transport protocols. UDP is preferred.

14.1.1 Related Terms

This section depicts some terms related to SIP.

Call
A call consists of all participants in a conference invited by a common source. A SIP call is
identified by a globally unique Call-ID.
Therefore, if several people invite a user to the same multicast session, each of these
invitations will be a call. A point-to-point Internet telephony conversation maps into a single
SIP call.

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Call Leg
The combination of Call-ID, To, and From identifies a call leg. For example, for a call
between A and B, the requests sent from A to B and from B to A use the same Call-ID,
belonging to the same call leg. A call leg is actually a path of messages for a call.

Transaction
A SIP transaction occurs between a client and a server. It comprises all messages from the
first request sent from the client to the server up to a final (non-1xx) response message sent
from the server to the client.
The CSeq sequence number within a call leg identifies a transaction. However, an ACK
request has the same CSeq number as the corresponding INVITE request, but comprises a
transaction of its own.
A normal call includes three transactions. Call initiation consists of two requests: INVITE and
ACK. The former requires a response. The latter is an acknowledgement that the final
response is received, requiring no response. Call termination contains one request BYE.

Location Service
Location services are offered by location servers. A SIP redirect or proxy server uses a
location service to obtain the possible location of a callee beyond the scope of this document.
However, the manner in which a SIP server requests location services is beyond the scope of
this manual.

Proxy Server
A proxy server is an intermediary program. It acts as both a server and a client to route SIP
requests to destinations. A proxy server may process requests internally or pass them on to
other proxy servers. It interprets, and, if necessary, rewrites a request before forwarding the
message.

Redirect Server
A redirect server performs the following:
z Accepts a SIP request.
z Maps the address into zero or more new addresses.
z Returns these addresses to the client.
Thus, the client can directly initiate requests to these new addresses again.
A redirect server implements the routing function instead of receiving or rejecting calls.

Registrar
A registrar is a server that accepts REGISTER requests. It is co-located with a proxy or
redirect server. A registrar needs to store the address mapping relationship in REGISTER
requests in a database for subsequent call processes. It can offer location services.

User Agent
A user agent is a logical entity that initiates or receives SIP requests.

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14.1.2 SIP Addressing

Uniform resource locators (URLs) are used within SIP messages to indicate the originator
(From), current destination (Request-URI), and final recipient (To) of a SIP request, and to
specify redirection addresses (Contact).
Commonly, a SIP URL has a syntax as follows:
SIP:user:password@host:port;transport-param|user-param|method-param|
ttl-param|maddr-param|other-param

The components of a SIP URI have the following meanings:

z SIP: indicates that SIP is used for the communication with a specified end system.
z user: consists of any characters in a form of e-mail address or telephone number.
z password: can be included in a SIP URL. However, the use is not recommended, because
the passing of authentication information in texts is a security risk.
z host: can be a host domain name or an IP address.
z port: indicates the port number to which a request is sent. The default is 5060, a public
SIP port number.
z transport-param: indicates which transport protocol to be used, TCP or UDP. The default
is UDP.
z user-param: can be a telephone number. A special function of SIP URL is to allow the
host to be an IP telephony gateway with a telephone number as the username. Two
values are available for this field: IP and phone. When the field is set to "phone", the
username is a telephone number and the corresponding end system is an IP telephony
gateway.
z method-param: specifies methods or operations to be used.
z ttl-param: designates the time-to-live (TTL) of a UDP multicast data packet. This
parameter is valid only when the Transport parameter is "UDP" and the maddr parameter
is "multicast address".
z maddr-param: provides the server address to be contacted for a user, overriding the
address supplied in the host field. This address is typically a multicast address.

The following parameters are optional:

z transport-param
z user-param
z method-param
z ttl-param
z maddr-param
z other-param

Currently, CSOFTX3000 supports SIP URLs in the format of E.164 number@IP address:port.
For example:
Sip:8613301080001@127.0.0.1:5060;

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14.2 SIP Message Types

SIP messages are encoded in a text form. There are two types of SIP messages: request and
response.

14.2.1 Request Messages

Request messages are sent from a client to a server to activate a specific operation. They are
INVITE, ACK, BYE, CANCEL, REGISTER, and OPTIONS. Table 14-1 describes the
functions of these messages.

Table 14-1 Request messages

Message Function

INVITE Initiate a session request to invite a user to participate in a session.

ACK Confirm that the client receives a final response to an INVITE request.
ACK messages are used only with INVITE message.
BYE Indicate that the client wishes to release a call already set up.
CANCEL Cancel a pending request.
CANCEL messages do not affect a completed request. A request is
considered completed if the client receives a final response from the
server.
REGISTER Register an address with a SIP server.
OPTIONS Query a server about its capabilities.

14.2.2 Response Messages

After receiving and interpreting a request message, a server returns a response message to a
client.
There are two types of response messages:
z Provisional response
Provisional responses indicate progress, but do not terminate SIP transactions.
z Final response
The 1xx responses are provisional, whereas other responses are considered final.
− 1xx: informational — a 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 — a SIP server fails to fulfill an apparently valid request.
− 6xx: global failure — the request cannot be fulfilled at any SIP server.

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Table 14-2 describes common 1xx and 2xx response messages.

Table 14-2 Response messages

Status code Message function

100 Try.
180 Ring.
181 Calls are being forwarded.
182 Queue.
183 A reliable provisional response.
200 OK.

14.3 SIP Message Structure

Both request and response messages consist of the following:
z A start line
The start line of a request message is a Request-Line, whereas that of a response message
is a Status-Line.
z One or more header fields
Header fields include general-header, request-header, response-header, and entity-header.
The header fields contain different parameters.
− General header fields
Accept | Accept-Encoding | Accept-Langrage | Call-ID | Contact | Cseq | Date |
Encryption | From | Record-Route | Require | Supported | Timestamp | To |
User-Agent | ViaRequest header fields
Authorization | Contact | Hide | Max-Forwards| Organization | Priority |
Proxy-Authorization | Proxy-Require | Route | Require | Response-Key | Subject
− Response header fields
Proxy-Authenticate | Retry-After | Server | Unsupported | Warning |
WWW-Authenticate
− Entity header fields
Allow | Content-Encoding | Content-Length | Content-Type | Expires
z An empty line indicating the end of the header fields
z An optional message body
A message body can use session description protocol (SDP) as the description format of a
session. In addition, a message body can encapsulate ISDN user part (ISUP) messages.

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14.3.1 Request Message Structure

Figure 14-1 shows the structure of a SIP request message, consisting of a request line,
message header, a message header, an empty line, and a message body. A carriage-return
line-feed (CRLF) distinguishes each parameter line in the message header.

Figure 14-1 Structure of a SIP request message

Status
Method Request-URI SIP-Version
line
Call-ID: value

From: value

To: value

Cseq: value

Via: value Message

header
Contact: value

Max-Forwards: value
Content-Length: value
Content-Type: value
.......

CRLF
Message
SDP
body

The Request-Line begins with a method token, followed by the Request-URI identifying the
peer URI and the SIP-Version identifying the protocol version, and ending with a CRLF.
Single space (SP) characters separate the elements.
Methods contain the following request message names:
z INVITE
z ACK
z OPTIONS
z BYE
z CANCEL
z REGISTER
The message header of a request message can be a general header, a request header, or an
entity header. The order of message header parameters is not fixed. Each parameter consists of
its name followed by a colon and a value. The value and colon are separated by a space.
A message header ends with a CRLF, followed by a message body.

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Figure 14-1 only shows some parameters possibly contained in the message header of a request
message.

Parameter Descriptions
The following describes some common parameters in the message header of a request
message.
z Call-ID
This field uniquely identifies a SIP call. A Call-ID is generally in such a format as
Call-ID:local-id@host
The "host" is a host domain name or an IP address. The "local-id" is a unique identifier
within "host". Call-IDs are case-sensitive.
z From
Request and response messages must contain a From general-header field, indicating the
initiator of a request message. A server copies the field from a request message to its
response message. This field is generally in such a format as
From: display-name <SIP-URL>;tag=xxxx
The "display-name" is characters rendered by a human-user interface. The system must
use the display name "Anonymous" if the identity of the client is to remain hidden.
The "tag" may appear in the From field of a request message. It must be present when
two instances of a user sharing a SIP address make call invitations with the same Call-ID.
The "tag" value must be globally unique. The user maintains the same tag throughout the
call identified by the Call-ID.
z To
This field specifies the recipient of a request message, with the same syntax as the From
field. Request and response messages must contain a To general-header field.
In SIP, the combination of Call-ID, From, and To fields identify a call leg.
z Command sequence (Cseq)
Clients must add the CSeq general-header field to every request message. A CSeq field
in a request contains the request method and a decimal sequence number unique within a
Call-ID.
The initial value of a sequence number is arbitrary. Consecutive request messages that
differ in request methods, headers, or bodies but have the same Call-ID must contain
monotonically increasing and contiguous sequence numbers. Retransmissions of the
same request message carry the same sequence number.
A server copies the CSeq value from a request message to its response message.
ACK and CANCEL request messages must contain the same CSeq value as that in the
corresponding INVITE request message, whereas a BYE request message must have a
higher sequence number.
A server remembers the highest sequence number for any INVITE request message with
the same Call-ID value. The server responds to, and then discards, any INVITE request
message with a lower sequence number.
z Via
This field is generally in such a format as
Via:sent-protocol sent-by;via-params comment
Where,

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sent-protocol=protocol-name / protocol-version / transport,via-params=via-hidden

| via-ttl | via-maddr | via-received | via-branch.
The Via field indicates the path taken by a request message so far. This prevents request
looping and ensures response messages take the same path as its request messages,
which assists in firewall traversal and other unusual routing situations.
The client originating a request message must insert into the request message a Via field
containing its host name or network address, and the port number at which it wishes to
receive response messages if it does not use the default port number.
In the process of sending a request onwards, each proxy server must add its own address
as a new Via field before any existing Via fields.
A proxy sending a request to a multicast address must add the "maddr" parameter to its
Via field. If a server receives a request message containing an "maddr" parameter in the
topmost Via field, the server will transfer the request message to the multicast address
listed in the "maddr" parameter.
Normally, every host that sends or forwards a SIP message adds a Via field indicating the
path traversed. However, network address translators (NATs) may change the source
address and port number of a request message. In this case, the Via field cannot be relied
on to route response messages.
To prevent this, a proxy server must check the top-most Via field. If the value of Via
field mismatches the previous hop address seen from the proxy server, the proxy adds a
"received" parameter to the Via field inserted by the previous hop. For example:
Via:SIP/2.0/UDP softx3000.bell-telephone.com:5060
Via:SIP/2.0/UDP 10.0.0.1:5060;received=191.169.12.30
In this example, the message originates from 10.0.0.1 and traverses a NAT with the IP
address 199.172.136.3 to reach the proxy server softx3000.bell-telephone.com. The
proxy sever does the following:
1. Notices the mismatch.
2. Adds a "received" parameter to the Via field of previous hop, containing the
address that the packet actually came from.
3. Appends its own address at the top as a new Via field.
A proxy server or a client processes the Via field in a response message according to the
following rules:
The first Via field should indicate the proxy or client processing this response message.
If it does not, discard the message. Otherwise, delete this Via field.
If there is no second Via field, this response message reaches its destination. Otherwise,
the processing depends on whether the Via field contains a "maddr" or a "receiver"
parameter:
If the second Via field contains a "maddr" parameter, send the response message to the
multicast address listed there, using the port indicated in "sent-by", or port 5060 if none
is present. The TTL of the response message should be the value indicated in the "ttl"
parameter. If that parameter is not present, set it to "1".
If the second Via field contains a "received" but not a "maddr" parameter, send the
message to the address indicated in the "received" parameter.
If neither of the previous cases applies, send the message to the address indicated by the
"sent-by" value in the second Via field.
An example of the Via field is:
Via: SIP/2.0/UDP 191.169.1.116:5061;ttl=16;maddr=191.169.10.20;branch=a7c6a8dlze
z Contact

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This field can appear in INVITE, ACK, and REGISTER request messages, and in 1xx,
2xx, and 3xx response messages. It provides a URL where a user can be reached for
further communications.
INVITE and ACK request messages contain Contact fields indicating from which
location a request message originates. This allows the called subscriber to send future
request messages, such as BYE, directly to the caller rather than through a series of
proxy servers.
This field is generally in such a format as
Contact:name-addr;q=value;action= “proxy” |
“redirect”;expires=value;extension-attribute
The "name-addr" form is the same as that in the To and From fields. The "q" value
ranges from 0 to 1. Higher values indicate higher preference. The "action" parameter is
only applicable to REGISTER request messages. It indicates whether a server expects
future requests to the client for the server proxy or redirection service.
If this parameter is not specified, the action taken depends on server configuration. The
"expires" parameter indicates how long a uniform resource identifier (URI) is valid. This
parameter can be a number indicating seconds or a quoted string containing a SIP-date.
The "extension-attribute" is an extension name.
An example of the Contact field is:
Contact: <Sip:66500002@191.169.1.110:5061>;q=0.7;expires=3600
z Max-Forwards
This field limits the number of times for which a request message is allowed to be
forwarded.
Each proxy server or gateway recipient of a request message containing a Max-Forwards
field must check and update the value of the field before forwarding the request. The
initial value is 70. It is subtracted by 1 every time a request message traverses a proxy
server or gateway.
If the received value is zero (0) and the request message does not reach its destination
address, the server returns 483 (too many hops) and terminates this request.
The purpose of setting this field is to prevent consuming proxy server resources in the
case of loop during message transfer.
This field is generally in such a format as
Max-Forwards: decimal integrals
z Allow
This field lists the set of methods supported by proxy servers.
An example of the Allow field is:
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE
z Content-Length
This field indicates the size of a message-body, in decimal number of octets.
Applications use this field to indicate the size of a message-body to be transferred.
If TCP serves as the transport protocol, the message header must contain this field.
An empty line separating a message header and a message body is beyond the scope of
the Content-Length. Any Content-Length greater than or equal to zero is a valid value. If
no body is present in a message, the Content-Length header field must be set to zero.
This field is generally in such a format as
Content-Length: decimal number of octets
z Content-Type

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This field indicates the media type of the message-body sent to the recipient. If the
message-body is not null, it must contain the Content-Type field.
An example of the Content-Type field is:
Content-Type: application/sdp
z Expires
This field gives the time after which the message content expires.

Request Message Examples

An example of SIP request message is:
INVITE sip:66500002@191.169.1.110 SIP/2.0
From: <sip:44510000@191.169.1.116>;tag=1ccb6df3
To: <sip:66500002@191.169.1.110>
CSeq: 1 INVITE
Call-ID: 20973e49f7c52937fc6be224f9e52543@sx3000
Via: SIP/2.0/UDP 191.169.1.116:5061;branch=z9hG4bkbc427dad6
Contact: <sip:44510000@191.169.1.116:5061>
Max-Forwards:70
Content-Length:230
Content-Type: application/sdp

v: 0
o: HuaweiSoftX3000 1073741831 1073741831 IN IP4 191.169.1.116
s: Sip Call
c: IN IP4 191.169.1.95
t: 0 0
m: audio 30000 RTP/AVP 8 0 4 18
a: rtpmap:8 PCMA/8000
a: rtpmap 0 PCMU/8000
a: rtpmap 4 G723/8000
a: rtpmap 18 G729/8000

The following details the example:

z Line 1: Request-Line
The Method is INVITE, followed by the Request-URI "sip:66500002@191.169.1.110"
and SIP-Version "2.0".
z Line 2: From field
It indicates the address of the request initiator is "sip:44510000@191.169.1.116".
"tag" is "1ccb6df3", differentiating the users who share a SIP address and make call
invitations with the same Call-ID.
z Line 3: To field
It specifies the address of the request recipient is "sip:66500002@191.169.1.110".
z Line 4: Cseq field
z Line 5: Call-ID field
It is globally unique, identifying a specific invitation.
z Line 6: Via field
It indicates the path taken by the request. "SIP/2.0/UDP" specifies the sent protocol, in
which "SIP" is the protocol name, "2.0" is the protocol version, and "UDP" is the
transport layer. "191.169.1.116:5061" indicates that the CSOFTX3000 IP address of the

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sender is "191.169.1.116" and port number is "5061". "branch=z9hG4bkbc427dad6" is

the "branch" parameter, identifying branches when the CSOFTX3000 distributes request
messages concurrently.
z Line 7: Contact field
It indicates subsequent request messages such as BYE can be directly sent to
"sip:44510000@191.169.1.116:5061" rather than through the Via field.
z Line 8: Max-Forwards field
It specifies that the maximum number of intermediate proxy servers or gateways that the
request message is allowed to traverse is 70.
z Line 9: Content-Length field
It indicates the length of the message body.
z Line 10: Content-Type field
z It indicates the message contains a single message body SDP.
z Line 11: an empty line
It indicates that the message header ends and below is the message body described by the
SDP.
The following briefs the message body. For details, see SDP-related documents.
v=<protocol version>
o=<user name><session ID><version><network type><address type><address>
s=<subject>
c=<network type><address type><connection address>
t=<start time><end time>
m=<media><port><transport layer><format list>
a=rtpmap:<payload type><encoding><code>

14.3.2 Response Message Structure

Figure 14-2 shows the structure of a SIP response message, consisting of:
z A status line
z A message header
z An empty line
z A message body
A line feed character distinguishes each parameter line in the message header. Parameters
vary with response messages.

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Figure 14-2 Structure of a SIP response message

SIP- Reason- Status

Status-Code
Version Phrase line
Call-ID: value

From: value
To: value

Cseq: value

Via: value Message

header
Contact: value

Max-Forwards: value
Content-Length: value
Content-Type: value
.......

CRLF
Message
SDP
body

A Status-Line consists of a SIP protocol version, a Status-Code, and its associated textual
phrase (Reason-Phrase). The Status-Code is a 3-digit integer code that indicates the type of a
request message. The Reason-Phrase gives a short textual description of the Status-Code.
The Status-Code includes 1xx, 2xx, 3xx, 4xx, 5xx, and 6xx, respectively defining six types of
SIP response messages. For full definitions, see section 14.2.2 "Response Messages".
The parameters in the header of a response message are the same as those in a request
message header. For details, see section 14.3.1 "Request Message Structure".

14.4 Introduction to SIP-T

The session initiation protocol for telephones (SIP-T) is an extension of SIP. It is a set of
mechanisms for encapsulating ISUP signaling within SIP. The purpose of SIP-T is to provide
better PSTN-SIP interconnection. There are three call models: PSTN-IP, IP-PSTN, and
PSTN-IP-PSTN.
SIP-T adopts SIP message structures and flows. Two techniques, namely, encapsulation and
mapping apply to SIP-T.
Encapsulation means that SIP message bodies contain ISUP messages, including two cases:
z A SIP message does not carry SDP. ISUP messages are encapsulated within the SIP
message body, which type is Application/ ISUP.
z A SIP message that carries SDP contains multiple message bodies. The type of the
message body with ISUP encapsulated in it is Application /ISUP.

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Mapping includes:
z ISUP-SIP message mapping
Gateways must generate a specific ISUP message for each SIP message received.
Conversely, gateways can also generate a specific SIP message for each ISUP message
received. SIP-T specifies the rules that govern the mapping between ISUP and SIP
messages. For example, an IAM must be sent on receipt of an INVITE, a REL for BYE,
and so on.
A mapping between ISUP and SIP messages can be as follows:
IAM = INVITE
ACM = 180 RINGING
ANM = 200 OK
REL = BYE
RLC = 200 OK for BYE
z ISUP parameter-SIP header mapping
A SIP request message that is used to set up a call contains information that enables the
message to be correctly routed to its destination, for example, a called number. SIP-T
defines the procedure for mapping of information from ISUP to SIP. For example, the
called number in an ISUP IAM must be mapped onto the SIP "To" header field.

14.5 SIP Signaling Procedures

This session does not describe the procedure that the CSOFTX3000 controls media gateways (MGWs)
to set up bearers.

14.5.1 Flows of Mobile Originated Calls Through SIP Trunks

The following describes the procedure of a mobile originated call (MOC) through the SIP
trunks between CSOFTX3000 devices. Figure 14-3 shows the flowchart.

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Figure 14-3 SIP flowchart of an MOC through SIP trunks

BSS CSOFTX3000 CSOFTX3000

CM_SERVICE_REQ

ASS_REQ

ASS_CMP
INVITE

100 Trying

180 Ringing

200 OK

ACK

CLEAR_REQ

BYE
CLEAR_CMD

200 for BYE

CLEAR_CMP

1. An ordinary mobile subscriber initiates a CM_SERVICE_REQ.

2. The CSOFTX3000 sends an ASS_REQ to the BSS.
3. The BSS returns an ASS_CMP.
4. After number analysis, the CSOFTX3000 finds that the outgoing office interacts with the
ingoing office through SIP. At the time, the CSOFTX3000 sends the called office an
INVITE containing the calling bearer information in the message body.
5. The called office returns the CSOFTX3000 a 100 Trying, indicating that it receives the
request message and message processing is in progress.
6. The CSOFTX3000 receives a 180 Ringing.
7. If the called subscriber hooks off, the called office sends the local CSOFTX3000 a 200
OK, containing the called bearer information in the message body.
8. After the MGW fulfils bearer setup under the control of the local CSOFTX3000, the
local CSOFTX3000 sends the peer office an ACK, indicating setup of the signaling path.

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9. If the calling subscriber hooks on, the CSOFTX3000 receives a CLEAR_REQ. At the
time, it sends a BYE to the called office and a CLEAR_CMD to the BSS at the calling
side. On receipt of the BYE, the called office sends a 200 for BYE to CSOFTX3000.
The BSS sends a CLEAR_CMP to confirm the completion of calling party disconnect.
10. If the called subscriber hooks on, the called office sends a BYE to the local
CSOFTX3000. On receipt of the BYE, the local CSOFTX3000 returns a 200 for BYE.
Meanwhile, it sends a CLEAR_CMD to the BSS at the calling side. The BSS returns a
CLEAR_CMP.

14.5.2 SIP-T Signaling Procedure

The following illustrates the call flow of transparency across points of PSTN-SIP
interconnection in terms of the PSTN-IP-PSTN call model. Figure 14-4 shows the flowchart.

Figure 14-4 A successful SIP-T procedure (PSTN-IP-PSTN)

LS A CSOFTX3000 A CSOFTX3000 B LS B

IAM
INVITE
IAM
100 Trying

ACM
180 Ringing
ACM ANM
200 OK
ANM
ACK

REL
BYE
REL

RLC
200 OK
RLC

1. A PSTN subscriber dials, LS A sends an IAM to CSOFTX3000 A.

2. CSOFTX3000 A preserves the received IAM in an INVITE message body that it sends
to CSOFTX3000 B.
3. CSOFTX3000 B extracts the IAM from the INVITE and sends the IAM to LS B.
4. CSOFTX3000 B returns CSOFTX3000 A a 100 Trying, indicating that it receives the
request message and message processing is in progress.
5. The called PSTN phone rings. Meanwhile, LS B sends an ACM to CSOFTX3000 B.
CSOFTX3000 B preserves the received ACM in a 180 Ringing that sends to
CSOFTX3000 A.

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6. CSOFTX3000 A extracts the ACM from the received 180 Ringing and forwards the
ACM to LS A. LS A receives the ACM. The calling PSTN subscriber hears ring back
tones.
7. If the called PSTN subscriber hooks off, LS B sends an ANM to CSOFTX3000 B.
CSOFTX3000 B preserves the received ANM in a 200 OK that it sends to CSOFTX3000
A.
8. CSOFTX3000 A extracts the ANM from the received 200 OK and forwards the ANM to
LS A.
9. CSOFTX3000 A sends an ACK to CSOFTX3000 B, acknowledging receipt of the final
message from CSOFTX3000 B in response to the INVITE.
10. At this time, both parties can communicate through an established bidirectional signaling
path.
11. If the calling PSTN subscriber hooks on, LS A sends a REL to CSOFTX3000 A.
CSOFTX3000 A preserves the received REL in a BYE message body that it sends to
CSOFTX3000 B.
12. CSOFTX3000 B extracts the REL from the received BYE and forwards the REL to LS
B.
13. On receipt of the REL, LS B sends busy tones to the called PSTN subscriber. If the
calling PSTN subscriber hooks on, LS B sends a RLC to CSOFTX3000 B.
14. CSOFTX3000 B preserves the received RLC in a 200 OK message body that it sends to
CSOFTX3000 A.
15. CSOFTX3000 A extracts the RLC from the received 200 OK and forwards the RLC to
LS A.

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