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kamailio/modules/seas/README

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Seas Module
Elias Baixas
VozTelecom Sistemas
www.wesip.eu
Ronda Can Fatjo, 9, 1p Parc Tecnologic del Valles Cerdanyola, 08520 (SP
AIN)
Phone:+34 933968800
www.voztele.com
<elias.baixas@voztele.com>
Copyright © 2006 VozTelecom Sistemas
__________________________________________________________________
Table of Contents
1. The Sip Express Application Server User's Guide
1. Application Servers
1.1. Sip Express Application Server module overview
1.2. Application Servers
1.3. Dependencies
1.3.1. Kamailio Modules
1.3.2. External Applications
1.4. Parameters
1.4.1. listen_sockets (string)
1.5. Functions
1.5.1. as_relay_t(String name)
1.5.1.1. Return value
2. WeSIP Application Server
2.1. The Servlet programming paradigm: Sip/Http Servlets
2.1.1. Converged Http/Sip Servlet Containers
2.2. Configuring WeSIP to work with SEAS
2.2.1. Server
2.2.2. Service
2.2.3. Connector
2.2.4. Engine
2.2.5. Mapper
2.2.6. Realm
2.2.7. Host
2.2.8. Mapper
2.3. Configuration Examples
2.3.1. kamailio.cfg in standalone
2.3.2. kamailio.cfg working as WeSIP front-end
2.3.3. Server.xml
2. Developer Guide
1. Internals
2. SEAS Protocol
2.1. The SEAS protocol
2.2. General codification of a header
2.2.1. Codification of a generic URI
2.2.2. Codification of To and From headers
2.2.3. Codification of Contact
2.2.4. Codification of Route and Record Route headers
2.2.5. Codification of Accept and Content-Type headers
2.2.6. Codification of Authorization headers
2.2.7. Codification of Allow headers
2.2.8. Codification of Content-Disposition headers
2.2.9. Codification of Content-Length header
2.2.10. Codification of Cseq header
2.2.11. Codification of Expires header
2.2.12. Codification of a SIP message
2.2.12.1. The general message information
section
2.2.12.2. The headers index section
List of Figures
1.1. SipServlet UML diagram
2.1. Overview of Seas Event Dispatcher process operation
2.2. SIP Messages and control flow within SER
2.3. General codification of a SIP header in SEAS protocol
2.4. Example of a from header SEAS-protocol codification
2.5. SEAS-codification of a SIP URI (byte meanings are shown)
2.6. Example of a SEAS SIP URI codification
2.7. SEAS codification of From and To headers
2.8. SEAS codification of a Contact header
2.9. SEAS codification of a Route Header
2.10. SEAS codification of Authentication/Authorization headers
2.11. SEAS codification of a SIP First Line
2.12. SEAS Headers Index section overview
2.13. SEAS SIP-Message codification
2.14. Different kinds of SEAS codified Events and Actions
List of Examples
1.1. Set listen_sockets parameter
1.2. as_relay_t usage
1.3. Typical example of an HttpServlet
1.4. Typical Sip Servlet Example
1.5. Server
1.6. Service
Chapter 1. The Sip Express Application Server User's Guide
Table of Contents
1. Application Servers
1.1. Sip Express Application Server module overview
1.2. Application Servers
1.3. Dependencies
1.3.1. Kamailio Modules
1.3.2. External Applications
1.4. Parameters
1.4.1. listen_sockets (string)
1.5. Functions
1.5.1. as_relay_t(String name)
1.5.1.1. Return value
2. WeSIP Application Server
2.1. The Servlet programming paradigm: Sip/Http Servlets
2.1.1. Converged Http/Sip Servlet Containers
2.2. Configuring WeSIP to work with SEAS
2.2.1. Server
2.2.2. Service
2.2.3. Connector
2.2.4. Engine
2.2.5. Mapper
2.2.6. Realm
2.2.7. Host
2.2.8. Mapper
2.3. Configuration Examples
2.3.1. kamailio.cfg in standalone
2.3.2. kamailio.cfg working as WeSIP front-end
2.3.3. Server.xml
1. Application Servers
1.1. Sip Express Application Server module overview
1.2. Application Servers
1.3. Dependencies
1.3.1. Kamailio Modules
1.3.2. External Applications
1.4. Parameters
1.4.1. listen_sockets (string)
1.5. Functions
1.5.1. as_relay_t(String name)
1.5.1.1. Return value
1.1. Sip Express Application Server module overview
SEAS module enables Kamailio to transfer the execution logic control of
a sip message to a given external entity, called the Application
Server. When the Kamailio script is being executed on an incoming SIP
message, invocation of the as_relay_t() function makes this module send
the message along with some transaction information to the specified
Application Server. The Application Server then executes some
call-control logic code, and tells Kamailio to take some actions, ie.
forward the message downstream, or respond to the message with a SIP
repy, etc.
The module acts implements a network protocol acting as the interface
between Kamailio internal API and the external Application Server
entity.
There's only one relevant function, as_relay_t, exported by this
module. This function receives as a parameter the name of the
application server to which the message should be relaied. Every
message relaied to an Application Server is automatically associated to
a SIP transaction (a transaction is created for it). Just after the
message is relaied to the Application Server, the script stops its
execution on the message, because the control of message-processing is
now in the Application Server.
In the context of SEAS module, relaying a message to an App Server, is
_not_ done in SIP protocol, but in a special protocol by means of which
the SEAS module and the Application Server comunicate efficiently and
seamlessly. This procotol is specially designed so that a message
doesn't need to be parsed again once it arrives at the Application
Server. This protocol carries information regarding the internal
structure of the SIP message (to avoid reparsing) and also information
about the associated transaction (recall that invoking as_relay_t
indirectly calls t_newtran). This way, all the SIP-Transaction
machinery, and the SIP-Message parsing, is handled at the Kamailio
core, while the execution of the Application Logic is carried in the
Application Server.
The SEAS module and protocol provide a means by which an external
entity can utilize Kamailio as a transaction-stateful SIP-stack to act
on behalf of it. This means that this external entity (which we call
the Application Server) is notified whenever a SIP-Request enters
Kamailio, and this external entity can then order Kamailio to execute
some actions, either replying the request, or generating new UAC
transactions.
This version of SEAS works with VozTelecom's WeSIP Application Server.
This Application Server is a SipServlet JAVA Container.
1.2. Application Servers
When Kamailio starts and SEAS module is loaded, a new process is spawn
which listens on a server-socket (IP and port are specified as a
parameter in the config script). From then on, the Application Servers
can connect to that socket so that Kamailio can relay messages to them.
When an Application Server connects to the socket, it sends its name
through the socket, so every App Server is identified with a name.
Within the Kamailio script, invoking as_relay_t() receives a string as
a parameter, which specifies the name of an application server to which
the message has to be sent. If that concrete application server hasn't
already connected to the module, the function returns a negative value,
otherwise (the Application Server is connected), the message is relaied
to it.
1.3. Dependencies
1.3.1. Kamailio Modules
SEAS module relies on the Transaction Module (TM module) for operation.
1.3.2. External Applications
Using the SEAS module requires to have an Application Server running
and connected to a particular instance of Kamailio.
At the moment, the only Application Server that works with SEAS is
WeSIP Application Server, which can be downloaded from www.wesip.eu,
and used freely for non-comercial purposes.
1.4. Parameters
1.4.1. listen_sockets (string)
The listen_sockets string tells SEAS where to listen for incoming
connections of Application Servers. It has the form: "ip:port". SEAS
will open two server-sockets on that IP, at the specified port, and
another at port+1. Application Servers must be configured to connect to
that port.
In case this parameter is ommited, SEAS listens on the default IP which
Kamailio is using, and opens the ports 5080 and 5081 to listen for
Application Servers.
Example 1.1. Set listen_sockets parameter
...
modparam("seas", "listen_sockets","127.0.0.1:5080")
...
1.5. Functions
1.5.1. as_relay_t(String name)
Creates a new transaction (if it isn't already created) and sends the
SIP Request and transaction information to the Application Server
specified in the parameter. Every Application Server connected to
Kamailio through the SEAS module, must be identified with a different
name.
This function can be used within REQUEST_ROUTE.
Example 1.2. as_relay_t usage
...
if (!as_relay_t("app_server_1")) {
log("Error sending to app server");
t_reply("500","App Server not connected");
}
...
1.5.1.1. Return value
In case the Application Server is connected to Kamailio, the function
does _not_ return, the Application Server is now in charge of
processing the request, and it may then reply to the request, initiate
new transactions, or whatever the application being executed wants.
In case the Application Server identified by the string parameter
passed to as_relay_t() is not connected to Kamailio, the function
returns 0, so that the script can continue processing the request.
2. WeSIP Application Server
2.1. The Servlet programming paradigm: Sip/Http Servlets
2.1.1. Converged Http/Sip Servlet Containers
2.2. Configuring WeSIP to work with SEAS
2.2.1. Server
2.2.2. Service
2.2.3. Connector
2.2.4. Engine
2.2.5. Mapper
2.2.6. Realm
2.2.7. Host
2.2.8. Mapper
2.3. Configuration Examples
2.3.1. kamailio.cfg in standalone
2.3.2. kamailio.cfg working as WeSIP front-end
2.3.3. Server.xml
At the moment, the only Application Server known to work with SEAS is
WeSIP. You can download a copy from www.wesip.eu.
WeSIP is a converged Sip/Http Servlet Container.
2.1. The Servlet programming paradigm: Sip/Http Servlets
Servlets are pieces of code that encapsulate the logic of an
application. Servlets are deployed into an Application Server. Whenever
a user requests service, the Application Server processes the request,
and passes control to the servlet. The servlet then executes some
logic, may it be a query to a database, the execution of a business
process, the creation of customized content for the user, or whatever
the service programmer could imagine. When the servlet finishes the
execution, it creates a response and gives it back to the Application
Server, which is in charge of making it reach back to the user. The
Application Server implements the network protocol, it takes care of
everything needed for a proper communication between user and server,
so the servlet doesn't have to care about these things. The servlet
uses a set of resources from the Application Server, such as Session
management, service routing or chaining, and request/response header
composition.
In HttpServlets, a service programmer has to implement a method in a
JAVA class, which could be called doGet() or doPost(). Whenever an HTTP
request arrived at the server, one of these functions was called with
the request as a parameter, so the logic of the application was
executed over that particular request.
HttpServlet has been extensively used over the past years, in all kinds
of business and web services.
This is how a typical HttpServlet looks like:
Example 1.3. Typical example of an HttpServlet
public final class Hello extends HttpServlet {
protected void doGet(HttpServletRequest request,HttpServletResponse response)
throws IOException, ServletException
{
response.setContentType("text/html");
PrintWriter writer = response.getWriter();
writer.println("<html>");
writer.println("<head>");
writer.println("<title>Sample Application Servlet</title>");
writer.println("</head>");
writer.println("<body bgcolor=white>");
writer.println("<table border=\"0\" width=\"100%\">");
Enumeration names = request.getHeaderNames();
while (names.hasMoreElements()) {
String name = (String) names.nextElement();
writer.println("<tr>");
writer.println("<th align=\"right\">"+name+":</th>");
writer.println("<td>"+request.getHeader(name)+"</td>");
writer.println("</tr>");
}
writer.println("</table>");
writer.println("</body>");
writer.println("</html>");
}
}
The successor of HttpServlet for SIP networks, is the SipServlet API.
Making most of the success of HttpServlet, the SipServlet API follows
the same programming paradigm, so that SIP application programmers can
reuse their knowledge in the field.
SipServlet API works the same way as HttpServlet: an Application Server
implements a SIP Stack and executes all the complex protocol logic. It
receives and pre-processes the requests from the network, and at the
right moment, passes control to the servlet doXxx() method, where the
programmer implemented the application logic. Depending on what kind of
SIP Message it was, a method or another will be executed. For example,
if an INVITE is received, the doInvite() method will be invoked in the
servlet.
The application can then access all the parts of the request and do its
work. When the service has been executed, it passes control back to the
Application Server with a response, so that it can be forwarded to the
user, and the service be satisfied.
Sip Servlets can be used to implement basic SIP network functionalities
(such as Proxy or Registrar servers), but their true power emerges in
the implementation of value-added services, which greatly surpasses the
basic service functionality of plain SIP servers.
Examples of value-added services, are Virtual PBX or IPCentrex,
Attended call forwarding, Instant Messaging, etc.
This is the appearance a typical SipServlet:
Example 1.4. Typical Sip Servlet Example
public class ProxyServlet extends SipServlet {
protected void doInvite(SipServletRequest req) throws
ServletException, IOException
{
if (req.isInitial()) {
Proxy proxy = req.getProxy();
proxy.setRecordRoute(false);
proxy.setParallel(parallel);
proxy.setSupervised(supervised);
SipURI rrURI = proxy.getRecordRouteURI();
rrURI.setParameter("foo", "bar");
req.setContent("Method is INVITE", "text/plain");
proxy.proxyTo(uris);
} else {
log("re-INVITE");
}
}
protected void doAck(SipServletRequest req) throws
ServletException, IOException
{
log("doAck " + req.getRequestURI());
if (req.isInitial()) {
throw new ServletException("unexpectedly got initial ACK");
The servlet programming API is event-ridden: every time a request comes
into the Application Server (may it be an Http or SIP one), the
specific servlet is executed and the service provided within it.
It is a very straightforward way of programming services, and the
Servlet API provides very easy and powerful means to access information
about the SIP-session or Http-session, about the request or response,
about the state of the dialog, or whatever it is needed.
The application programmer has a rich framework of resources that allow
him to focus only on the service logic, without having to worry about
the underlying protocol specifics (SIP or HTTP).
Figure 1.1. SipServlet UML diagram
SipServlet UML diagram
The Servlet programming language is JAVA, which offers a wide spectrum
of programming API's dealing with all kinds of techniques, tools and
resources, which also are available seamlessly from the Servlet
context.
This makes the SipServlet API very desirable for all kinds application
developers.
SipServlet allows a rapid SIP application development and deployment,
and also provides a reliable and secure framework of service execution
(the JAVA sandbox and the Application Server execution environment).
2.1.1. Converged Http/Sip Servlet Containers
SipServlets achieve the most of it when they can be deployed along with
HttpServlets, in the same Application Server (also known as Servlet
Container). This environment truly realizes the power of converged
voice/data networks: Http protocol represents one of the most powerful
data transmission protocols used in modern networks (think of the SOAP
web-services protocol), and SIP is the protocol of choice in most of
the modern and future voice over IP (VoIP) networks for the signaling
part. So an Application Server capable of combining and leveraging the
power of these two APIs will be the most successful.
Convergence of SIP and HTTP protocols into the same Application Server
offers, amongst others, the following key advantages:
-It doesn't require to have 2 different servers (Http and Sip) so it
relieves from maintenance problems, and eases user and configuration
provisioning.
-It offers great convenience to the application programmer to have all
the classes related to the different protocols handled within the same
code.
-As it eases development of interactive and multimedia services,
realizing the power of well-known web-services and intermixing them
with new voice services.
These are some simple, but suggestive examples of services that could
be developed within a converged Http/Sip servlet:
-IP Centrex: through the use of the Web-interface, users could have a
layout of the office in a web page, and see what phones were ringing at
a given moment, so they could pick-up a call ringing in another phone
in their own desktop. Or they could forward a call to another party by
clicking on the web page and selecting which of the office phones it
had to be transferred to.
-Voicemail: users could upload an audio file to the server through a
web-page, to be used as the automatic answering message, and then also
download their voicemail through the web-page, or organize the messages
and remove the old ones.
-Instant Messaging: users could continue a voice call by starting or
joining a new Instant Messaging session carried over a web-page.
-Click-to-dial: users could initiate SIP sessions only by clicking a
link on a web page, without the need of the Web-Browser being SIP-aware
nor needing even a SIP phone: the server could handle all the logic so
the user who clicked could receive a call from the server's SIP
network.
2.2. Configuring WeSIP to work with SEAS
The WeSIP Application Server configuration file is based on the Apache
Tomcat configuration system: It is an XML-formatted file, in which the
different components of the server are specified.
The default config file that comes with the WeSIP distribution package
should be suitable for most of the deployment configurations.
2.2.1. Server
The topmost element in the XML configuration file is the "server" which
has 2 xml attributes, called "port" and "shutdown". The former
specifies a port on which the WeSIP AS will listen for the shutdown
command, and the latter is the magic word that will make the server
shutdown.
Example 1.5. Server
<Server port="8005" shutdown="SHUTDOWN" >
if you send the magic word "SHUTDOWN" to the port 8005 of the
localhost, the server will stop cleanly.
2.2.2. Service
Nested within the Server element, must be a "Service" element, with an
attribute called "name" which specifies the name for the service. This
attribute is not very relevant, you can call it whatever you like.
Example 1.6. Service
<Service name="WeSIP-Standalone">
Within the Service element must be two or more elements: the connectors
and the engines. A connector is the instance that will receive messages
from the network. You can specify HTTP connectors and/or SIP
connectors. Every connector needs an attribute called "className" which
specifies which class will be responsible for receiving the messages
from the network. For HTTP connectors, the classname must be
"org.apache.catalina.connector.http.HttpConnector" and for SIP
connectors "com.voztele.sipservlet.connector.SipConnector".
2.2.3. Connector
The SIP Connector uses 4 attributes:
className="com.voztele.sipservlet.connector.SipConnector"
specifies the classname of the connector.
minProcessors="5"
specifies the minimum number of SIPprocessor instances (and threads in
the pool) to process incoming SIP messages. More processors should
allow more load to be processed. This is the minimum number of
instances, even if they are spare and not working.
maxProcessors="10"
specifies the maximum number of SIP processors used (a negative value
specifies that there is no limit).
addresses="localhost:5060"
Specifies the SIP address and port in which the Application Server from
which the Application Server will process the SIP messages. This Addres
is where Kamailio listens for the messages, so in fact, Kamailio is
listening on them, but Kamailio passes the messages to WeSIP, so WeSIP
must be aware of this IP/port.
Warning
this attribute MUST match one of the listening points declared within
Kamailio in the "listen" parameters.
For example in kamailio.cfg:
listen = tcp:localhost:5060
listen = udp:localhost:5060
Within the SIP Connector element there must be an ExtraProperties
element, containing nestes Property elements. Each property element
specifies a parameter for the SIP Stack. Each property is specified by
a key and a value. The most significant keys are:
* com.voztele.javax.sip.SER_ADDRESS
This specifies the IP and port in which the Kamailio is listening
for Application Servers to connect and register.This specifies the
IP and port in which the Kamailio is listening for Application
Servers to connect and register.
Warning
This needs to match the listen_sockets seas module parameter within
the Kamailio configuration file. Ie.:
modparam("seas", "listen_sockets","127.0.0.1:5080")
* javax.sip.STACK_NAME
Specifies the name identifying this instance of the Application
Server.
Warning
This is the name you will set in the Kamailio configuration script
when you invoke the WeSIP Application Server, by calling the
as_relay_t function. This is the name you pass as the parameter of
the function. If you have different WeSIP instances all connecting
to the same Kamailio, they must each one have a different
STACK_NAME", and within Kamailio you can call each of them by
invoking as_relay_t() with a different name. Example:
<Property key="javax.sip.STACK_NAME" value="app_server_one" />
* com.voztele.javax.sip.THREAD_POOL_SIZE (integer)
Specifies the number of threads there must be in the pool to
process incoming SIP messages. If unspecificed, the default is
"infinity".
* com.voztele.javax.sip.SPIRAL_HDR
This property tells WeSIP and SEAS that every SipRequest and UAC
transaction generated from WeSIP, must spiral through SER, and will
be added a special Header called "X-WeSIP-SPIRAL: true" this will
make all the outgoing messages pass again through the Kamailio
script, so that they can be accounted or whatever the configurator
wants. For example, the configuration script could go:
route{
if(is_present_hf("X-WeSIP-SPIRAL")){
/* account, log, register, or whatever */
t_relay();
}else{
as_relay_t("app_server_1");
}
}
2.2.4. Engine
The Engine must also be nested within the Server element, along with
the Connectors. It must have a "name" attribute with whatever name you
feel like. It needs to have another attribute called "defaultHost"
which will be the default host to which to pass the incoming request
(in HTTP/1.0 the requests dont have a Host header, so they will be
passed to this default host, in SIP, this attribute doesn't have a
meaning.). In order to have this Engine handling also SIP messages, the
"className" attribute of the Engine must be
"com.voztele.sipservlet.core.ConvergedEngine".
Within the Engine, there can be one or more Hosts, each one specified
within a "Host" element nested in the engine.
2.2.5. Mapper
A mapper is used to map an incoming request to one or another SIP or
HTTP host. In case it is a SIP request, the mapping is done based on
the sip.xml deployment descriptor rules. The classname of the SIP
mapper MUST BE "com.voztele.sipservlet.core.EngineSipMapper". The
"mapper" element must also have a "protocol" attribute, specifying
which protocol this mapper handles. In case of the SIP mapper it must
be "SIP/2.0". The HTTP mapper's classname must be
"org.apache.catalina.core.StandardEngineMapper" and the protocol
attribute "HTTP/1.1"
2.2.6. Realm
The authentication in HTTP is performed in Apache-Tomcat through
Realms. The memory realm is (textual copy from the Apache-Tomcat
javadoc"): "Simple implementation of Realm that reads an XML file to
configure the valid users, passwords, and roles."
The classname must be "org.apache.catalina.realm.MemoryRealm"
A "pathname" attribute can be specified to tell the Realm which file
contains the usernames, passwords and roles. If not specified, it is
"conf/wesip-users.xml"
2.2.7. Host
A Host represents a VirtualHost in HTTP/1.1 servers, so the requests
will be dispatched to one or another virtual host depending on the
Host: header. In SIP this doesn't make much sense, because there's no
such Host: header, and virtual hosting is not done in this way. Every
host must have a "name" attribute which specifies the name of the
virtual host, it must also have a "nameSip" attribute which MUST MATCH
the IP or hostname _and_ port" specified in Kamailio listen parameters
and in the Sip Connector the hostname and the port must be separated
with an underscore. for example: nameSip="localhost_5060" or
nameSip="192.168.1.1_5060" The next important attribute that must have
the Host element is "appBase" which declares the directory where the
WEB and SIP applications reside. It usually is a directory called apps
in the directory from which the server runs. The attribute "unpackWARs"
says the WeSIP Application Server to unpack the Web or Sip Application
Archives (.war or .sar extensions) found inside the appBase directory.
It should usually be set to "true". The "port" attribute specifies the
port where this host is going to receive SIP messages . This only has
to do with the SIP protocol, not with HTTP. It must be the same as the
port specified in Kamailio parameter "listen_sockets" (for the seas
module). The "autoDeploy" attribute tells the host to monitor the
"appBase" directory for new application archives (.sar or .war) so they
can automatically be deployed. This parameter should be set to "true".
The "className" used for the Host _must_be_
"com.voztele.sipservlet.core.ConvergedHost"
2.2.8. Mapper
Hosts must also have a nested Mapper element, but when the mapper is
inside a Host (and not in an Engine) the classnames must be
"com.voztele.sipservlet.core.SipHostMapper" for the "SIP/2.0" protocol
and "org.apache.catalina.core.HttpHostMapper" for the "HTTP/1.1"
protocol. (2 mappers must be nested inside the Host).
2.3. Configuration Examples
In general, you can configure WeSIP to work with your Kamailio in two
ways: have 2 Kamailio instances, the first acting as
Proxy/Registrar/Redirect and the second cooperating with WeSIP to act
as the Application Server. This is the preferred deployment layout, as
the first Kamailio works as usual, and the requests that need special
services are relaied to another Kamailio which acts on behalf of the
WeSIP AS. This configuration profile distributes load (call-routing
logic in one instance, and Application Services in the other), and is
also more fault-tolerant. On the other hand, you can have all your
call-routing logic and Application Server on the same Kamailio, having
one script handle all the logic, and then invoking the App Server at
any point.
2.3.1. kamailio.cfg in standalone
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
loadmodule "/usr/local/lib/kamailio/modules/sl.so"
loadmodule "/usr/local/lib/kamailio/modules/tm.so"
loadmodule "/usr/local/lib/kamailio/modules/rr.so"
loadmodule "/usr/local/lib/kamailio/modules/maxfwd.so"
loadmodule "/usr/local/lib/kamailio/modules/usrloc.so"
loadmodule "/usr/local/lib/kamailio/modules/registrar.so"
loadmodule "/usr/local/lib/kamailio/modules/textops.so"
loadmodule "/usr/local/lib/kamailio/modules/seas.so"
loadmodule "/usr/local/lib/kamailio/modules/mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
modparam("usrloc", "db_mode", 0)
modparam("rr", "enable_full_lr", 1)
modparam("seas", "listen_sockets", "127.0.0.1:5080");
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
if (!method=="REGISTER")
record_route();
if (loose_route()) {
append_hf("P-hint: rr-enforced\r\n");
route(1);
};
if (uri==myself) {
if (method=="REGISTER") {
save("location");
exit;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
};
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
};
append_hf("P-hint: usrloc applied\r\n");
};
route(1);
}
route[1] {
if(!as_relay_t("app_server_one")){
t_reply("500","Application Server error");
}
}
2.3.2. kamailio.cfg working as WeSIP front-end
debug=9 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
reply_to_via=1
listen = tcp:localhost:5060
listen = udp:localhost:5060
mpath="/home/elias/src/sipservlet/seas"
loadmodule "modules/tm/tm.so"
loadmodule "modules/seas/seas.so"
loadmodule "modules/mi_fifo/mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
modparam("seas", "listen_sockets","127.0.0.1:5080")
route{
if(!as_relay_t("app_server_1")){
t_reply("500","Application Server error");
}
}
2.3.3. Server.xml
<Server port="8005" shutdown="SHUTDOWN" debug="0">
<Service name="WeSIP-Standalone">
<Connector className="org.apache.catalina.connector.http.HttpConnector"
port="8080" minProcessors="5" maxProcessors="75"
enableLookups="true" address="localhost" acceptCount="10" debug="10" />
<Connector className="com.voztele.sipservlet.connector.SipConnector"
minProcessors="5" maxProcessors="75"
addresses="localhost:5060" >
<ExtraProperties>
<Property key="com.voztele.javax.sip.SER_ADDRESS" value="
127.0.0.1:5080" />
<Property key="javax.sip.STACK_NAME" value="app_server_on
e" />
<Property key="com.voztele.javax.sip.THREAD_POOL_SIZE" va
lue="10" />
</ExtraProperties>
</Connector>
<Engine name="Standalone" defaultHost="localhost" debug="10"
className="com.voztele.sipservlet.core.ConvergedEngine">
<Logger className="org.apache.catalina.logger.SystemOutLogger"
timestamp="true"/>
<Mapper className="org.apache.catalina.core.StandardEngineMapper" protoc
ol="HTTP/1.1"/>
<Mapper className="com.voztele.sipservlet.core.EngineSipMapper" protocol
="SIP/2.0"/>
<Realm className="org.apache.catalina.realm.MemoryRealm" />
<Host name="localhost" nameSip="localhost_5060" debug="10" appBase="weba
pps" unpackWARs="true"
port="5060" autoDeploy="true" className="com.voztele.sipservlet.core.Con
vergedHost">
<Mapper className="com.voztele.sipservlet.core.SipHostMapper" pro
tocol="SIP/2.0"/>
<Mapper className="org.apache.catalina.core.HttpHostMapper" proto
col="HTTP/1.1"/>
</Host>
</Engine>
</Service>
</Server>
Chapter 2. Developer Guide
Table of Contents
1. Internals
2. SEAS Protocol
2.1. The SEAS protocol
2.2. General codification of a header
2.2.1. Codification of a generic URI
2.2.2. Codification of To and From headers
2.2.3. Codification of Contact
2.2.4. Codification of Route and Record Route headers
2.2.5. Codification of Accept and Content-Type headers
2.2.6. Codification of Authorization headers
2.2.7. Codification of Allow headers
2.2.8. Codification of Content-Disposition headers
2.2.9. Codification of Content-Length header
2.2.10. Codification of Cseq header
2.2.11. Codification of Expires header
2.2.12. Codification of a SIP message
2.2.12.1. The general message information section
2.2.12.2. The headers index section
1. Internals
The SEAS module runs within the Open Sip Express Router aka. Kamailio.
Kamailio uses a pool of processes to execute the script logic on every
new message received. These are called the worker processes. One of
these processes will be selected to process the script, and at some
point it will find a function invoking the relay of the SIP message to
one of the Application Servers registered. This function has been
called as_relay_t, which stands for Application Server relay (the _t
stands for TransactionStatefully), and receives as the only parameter
the name of the application server to be invoked.
The process will execute the as_relay_t function, which looks up in a
table if there is a registered Application Server with that name. If
there is one, the process will craft the SEAS header for the SIP
message being handled, put it in a shared memory segment, and write the
address of that segment to a pipe (4 bytes pointer in IA32).
This way, we will have all the Kamailio processes composing the SEAS
header along with the SIP message, and putting its shared memory
address into that pipe. This technique of inter-process communication
avoids race conditions because writing to a pipe is granted to be an
atomic operation if the data to write is less than _POSIX_PIPE_BUF,
which usually is 512 bytes.
At the initialization of Kamailio, the SEAS module creates the
discussed pipe, so that all the Kamailio worker processes inherit the
file descriptor associated to the pipe. Then it spawns a new process,
which will be the one to open two server sockets, and wait for the
Application Servers to connect and register.
Each Application Server wishing to receive events from Kamailio, will
have to open a socket to the module (the port and IP of the socket are
defined at start time in the script). After connection, it has to print
its identification name. The SEAS process (from now on, called event
dispatcher) will then register it in its internal structures, so that
the Kamailio processes can push events for it. The following picture,
shows the internals of the SEAS Event dispatcher process:
Figure 2.1. Overview of Seas Event Dispatcher process operation
Overview of Seas Event Dispatcher process operation
Within the SER server, the flowing of SIP Messages and control flow, is
depicted in the following diagram:
Figure 2.2. SIP Messages and control flow within SER
SIP Messages and control flow within SER
2. SEAS Protocol
2.1. The SEAS protocol
2.2. General codification of a header
2.2.1. Codification of a generic URI
2.2.2. Codification of To and From headers
2.2.3. Codification of Contact
2.2.4. Codification of Route and Record Route headers
2.2.5. Codification of Accept and Content-Type headers
2.2.6. Codification of Authorization headers
2.2.7. Codification of Allow headers
2.2.8. Codification of Content-Disposition headers
2.2.9. Codification of Content-Length header
2.2.10. Codification of Cseq header
2.2.11. Codification of Expires header
2.2.12. Codification of a SIP message
2.2.12.1. The general message information section
2.2.12.2. The headers index section
SIP is a very flexible protocol. It can be very easily extended with
new features, and SIP entities have a high level of freedom in
composing the SIP messages, for example setting IPs or hostnames in
URIs, reordering header fields, folding headers, aggregating/scattering
headers, etc.
This flexibility, though, makes it difficult to implement efficiently,
because parsing of text headers requires a lot of state.
Kamailio implements a very efficient parsing mechanism and
SIP-transaction machinery. The goal of the SEAS protocol is to keep all
this information that has been already extracted at Kamailio, so that
it can be reused at the Application Server.
2.1. The SEAS protocol
The SEAS protocol is a layer of information regarding the internal
structure of a SIP message that is added whenever SEAS sends a SIP
event to the Application Servers. The protocol is used for
communication between Kamailio and the Application Servers.
Once an incoming SIP message has reached the worker process within
Kamailio, it copies its content into a private memory area (which is, a
memory chunk not shared across processes). In this point, the message
first line is parsed to know whether it is a SIP request or response.
Kamailio uses a technique called lazy-parsing, which consists in
delaying the parse of headers until some piece of the code requires it.
As the SIP message goes traversing functions and the script code, a
function called parse_msg() gets called again and again, and the SIP
message gets parsed further and further. Each call to parse_msg passes
an integer value argument (32 bits) in which every bit signals a header
to be parsed, if they are already parsed (because a previous invocation
of parse_msg), the function returns immediately, otherwise, the SIP
message is scanned and parsed until all the headers requested get
parsed.
In each call to parse_msg, different parts of the message are analyzed,
and different SIP header-specific structures get filled. Each one of
this structures, give quick access to each of the parts of a SIP
message header.
For example, a Via header struct is called via_body, and has these
members: name, version, transport, host, proto, port, port_str, params,
comment, received, rport, etc. each of these members gives quick access
to each of the parts of the header. For example, a via header like
this: "Via: SIP/2.0/UDP 192.168.1.64:5070;branch=z9hG4bK-c02c60cc"
would have the member proto pointing to the "U" of "UDP", and a length
of 3, the host member would be pointing to "192.168.1.64" and have a
length of 12, the branch member would be pointing to "z9hG4bK-c02c60cc"
and a length of 16, and so on.
This structure is the result of the parsing. All this meta-information
regarding the SIP message structure, is stored in a sip_msg structure,
using dynamically-allocated memory segments.
Kamailio defines different structure types describing different SIP
headers, such as via_body, to_body, cseq_body, via_param, and so on.
These structures are generally composed of another kind of structure
called str.
The str structure is a key component of Kamailio's high performance. In
the C programming language, a string's length is known because a '0'
(null-character) is found at the end of it. This forces each of the
string manipulation functions to keep looking for a '0' in the byte
stream, which is quite processor consuming. Instead of this, Kamailio
defines a structure composed of a char pointer and an integer. The char
points to the start of a string, and the integer gives its length, thus
avoiding the '0' lookup problem, and giving a significant performance
boost.
This structure has been quite useful to the design of the SEAS
protocol, because it enables the description of the SIP message anatomy
by giving pointers to each of its fields, and integers describing each
of its lengths.
Knowing that a SIP header does not usually occupy more than a few
characters (always less than 256), the pointer in the structure has
been relativized to the beginning of the SIP message or the beginning
of the SIP header, and the integer giving the length, has been casted
to an unsigned byte (256 values, so 256 characters maximum length).
When messages get transferred from Kamailio to the Application Server,
it is optimum to keep this worthy meta-information regarding the SIP
message, so that it can be used at the AS part. For this to be
possible, it is needed to store the pointers to each of the syntactic
structures and their length.
In general, pointers are variables that point to a region in the memory
of a computer. The region of the memory is counted from the 0x00000000
address in IA32 architectures (from the beginning).
C provides functionality to do any kind of arithmetic operations over
pointers (add, subtract, multiply and divide), so that the euclidean
distance over the one-dimension address space can be calculated just by
subtracting a base address from another pointer.
These pointers will have to be transmitted through the network, along
with the SIP message, so for the pointers to keep their meaning, they
need to be relativized to a known point, and the most meaningful known
point in a SIP message is its start.
So making the pointers relative to the message start, gives two
important features: first, it makes the pointers still valid when they
arrive at another computer (because they are relative to the beginning
of the message), and they occupy far less memory, because from a 4-byte
pointer (in IA32) it gets translated to a 1 or 2 byte index, because an
important amount of redundant information is elicited (we already know
that each of the parts of the message belong to the message, so why
carry the message begin address in each of the pointers ?).
The SIP messages are composed of protocol headers and a payload. The
headers section don't usually surpass the 1500 byte limit, amongst
other reasons, because the usual Maximum Transmission Unit in Ethernet
networks is 1500 bytes and the protocol was initially designed to work
on UDP. For that reason, 11 bits should be enough to address a
particular region within the SIP message, because it yields 2048
positions. The closest greater value to 11 bits multiple of a byte (the
basic TCP network transport unit) is 16 bits, or 2 bytes, which makes
it possible to address 65536 positions from the beginning.
For the SEAS protocol to be extensible and platform-independent, all
the 2-byte pointers or indexes to each of the message regions are sent
in network-byte-order, or big endian. This is also useful in the JAVA
part to retrieve the indexes, because the JAVA natively uses a
big-endian representation of integers, regardless the architecture on
which it runs.
For each kind of standard SIP header (this is, the headers referred to
in the SIP specification) there is a code specification, regarding the
composition of the header. Each one of its parts points to one the
several components of the header. For example, a From header always has
a SipURI and may have several parameters, amongst others, a tag. Then,
the From header code has a field indicating where the URI starts, a
codification of the URI, and several pointers that point to each one of
the parameter names and values. This is the codification of the From
header. All the other headers have a similar codification.
2.2. General codification of a header
Every header codification, regardless it is known to the server or not,
begins with a 2-byte unsigned integer, which points to the beginning of
that header counted from the SIP message begin (a SIP message start
based pointer to the header). Following these two bytes is another byte
giving the length of the name, and another byte giving the length of
the entire header (including name and value).
Figure 2.3. General codification of a SIP header in SEAS protocol
General codification of a SIP header in SEAS protocol
For example:
Figure 2.4. Example of a from header SEAS-protocol codification
Example of a from header SEAS-protocol codification
2.2.1. Codification of a generic URI
As the SIP URI is one of the most used types in a SIP message, a
special structure has been defined to describe the contents of it. A
URI is always included inside a SIP header, or may be in the first line
of a SIP Request (as the request URI).
The codification of any URI is as follows:
Figure 2.5. SEAS-codification of a SIP URI (byte meanings are shown)
SEAS-codification of a SIP URI (byte meanings are shown)
What follows is an example of a SIP URI codification with the SEAS
protocol.
Figure 2.6. Example of a SEAS SIP URI codification
Example of a SEAS SIP URI codification
The first byte in the encoded-URI structure, gives the index where the
URI starts, counting from the beginning of the SIP header where it
appears. The next two bytes are flags indicating known fields present
in the URI (such as port, host, user, etc.).
All the following bytes are uri-start based pointers to the fields that
are present in the URI, as specified by the flags. They must appear in
the same order shown in the flags, and only appear if the flag was set
to 1.
The end of the field, will be the place where the following pointer
points to, minus one (note that all the fields present in a URI are
preceded by 1 character, ie
sip[:user][:passwod][@host][:port][;param1=x][;param2=y][?hdr1=a][&hdr2
=b]) it will also be necessary to have a pointer at the end, pointing
two past the end of the URI, so that the length of the last header can
be computed.
The reason to have the "other parameters" and headers flags at the
beginning (just after the strictly URI stuff), is that it will be
necessary to know the length of the parameters section and the headers
section. The parameters can appear in an arbitrary order, they won't be
following the convention of transport-ttl-user-method-maddr-lr, so we
can't rely on the next pointer to compute the length of the previous
pointer field, as the ttl parameter can appear before the transport
parameter. So the parameter pointers must have 2 bytes: pointer+length.
2.2.2. Codification of To and From headers
To and From headers follow the same structure, so the same codification
structure has been used to describe both. The structure is depicted in
the drawing:
Figure 2.7. SEAS codification of From and To headers
SEAS codification of From and To headers
2.2.3. Codification of Contact
The contact header is one of those SIP headers that can be combined,
which means that if several headers of the same type are present in the
message, they can be aggregated in a single header, having the header
values separated by a comma. Thus, a single Contact header can contain
more than one contact-value. For this reason, the Contact codification
is composed of a several Contact codifications concatenated, and a byte
at the beginning telling how much Contact codifications are present.
The code is depicted in the following drawing:
Figure 2.8. SEAS codification of a Contact header
SEAS codification of a Contact header
2.2.4. Codification of Route and Record Route headers
Both Route and Record-Route headers follow an identical structure, and
it is also permitted to combine several headers into one, with their
bodies (or header values) separated by commas. In this case, both kinds
of headers follow the same structure, defined as follows:
Figure 2.9. SEAS codification of a Route Header
SEAS codification of a Route Header
2.2.5. Codification of Accept and Content-Type headers
These two kinds of headers carry mime type and subtype definitions in
the form "type/subtype" (ie. text/xml, application/sdp or whatever).
For internal handling of this headers, SER codifies the known types and
subtypes into a single 32 bit integer, with the highest two bytes
giving the mime type, and the lowest two bytes giving the subtype.
The difference is that Accept header can also be combined, carrying
more than one header value in a single header row. Thus the Accept
header has a leading byte giving the number of mime type/subtype
integers present, while the Content-Type only uses 4 bytes (a 32-bit
integer) giving the type/subtype.
2.2.6. Codification of Authorization headers
SIP has inherited the authentication scheme from HTTP, which is based
on a digest scheme. There are several headers regarding these
authorization scheme, namely Proxy-Authenticate, WWW-Authenticate,
Authorization and Proxy-Authorization. All of them can be codified
using the same schema, which is as follows:
Figure 2.10. SEAS codification of Authentication/Authorization headers
SEAS codification of Authentication/Authorization headers
For each field present, there are 2 bytes, one pointing the place where
it starts, the next giving how long this field is. The URI is a special
case, and is composed of 1 byte telling how long is the URI structure,
and then the encoded URI structure.
2.2.7. Codification of Allow headers
Allow headers carry request methods that a user agent or proxy
understands or is willing to accept. In SER, request methods are
codified into a 32-bit integer, each of its bits signals a different
kind of header. The Allow header is codified copying that integer into
the payload of the header.
2.2.8. Codification of Content-Disposition headers
The content-disposition is encoded within 2 bytes: the first is a
header-start based pointer to where the content-disposition value
starts, and the second is its length. If there are parameters present,
each of them uses 1 byte pointing to where the parameter name starts,
and 1 byte pointing to where the parameter value starts. From these two
values, the parameter name and value lengths can be inferred.
2.2.9. Codification of Content-Length header
The content length header is codified as a 4-byte unsigned integer, in
network byte order.
2.2.10. Codification of Cseq header
The Cseq header is codified using 9 bytes. The first one is a number
corresponding to the internal value that SER assigns to that request
method (the method ID). The following 4 bytes are an unsigned 32-bit
integer according to the Cseq number. The next two bytes are the header
based pointer to the beginning of the Cseq number and its length, and
two more bytes pointing to the beginning of the method name and its
length.
2.2.11. Codification of Expires header
The expires header is composed of 6 bytes. The first four bytes are an
unsigned 32-bit integer with the parsed value of the header (which is
the number of seconds before a request expires). Then follows 1 byte
pointing to the beginning of the header value (the expires value as a
string) and a byte giving the length of the value.
2.2.12. Codification of a SIP message
2.2.12.1. The general message information section
In SER, not only the headers are parsed with a high degree of
optimization, but also the first line is. So for the SEAS protocol to
realize this improvement, a codification for the first line of every
SIP messages has also been defined.
The first two bytes of the codification are a 2-byte unsigned integer.
If its value is equal or greater than 100, then this is a response, and
the integer represents its status code. If its value is smaller than
100, then it is a request, and the integer represents the method of the
request being transported.
Figure 2.11. SEAS codification of a SIP First Line
SEAS codification of a SIP First Line
The next two bytes are an unsigned integer which is a pointer to where
the actual SIP message starts, beginning from the start of the codified
payload.
The next two bytes are also an unsigned integer giving the SIP message
length.
The next bytes differ on the meaning depending on whether the message
is a SIP Request or Response.
In case it is a Request:
The next two bytes, are a SIP-message-start based pointer to where the
method begins, and the method length.
The next two bytes, are a SIP-message-start based pointer to where the
Request URI begins, and the request URI length.
The next two bytes, are a SIP-message-start based pointer to where the
version identifier begins, and the version identifier length.
In case it was a Response:
The next two bytes, are a SIP-message-start based pointer to where the
response code begins, and the response code length.
The next two bytes, are a SIP-message-start based pointer to where the
reason phrase begins, and the reason phrase length.
The next two bytes, are a SIP-message-start based pointer to where the
version identifier begins, and the version identifier length.
In case the message is a SIP response, the following bytes correspond
to the Request URI codification. The first byte is the length of the
URI codification, followed by the URI code.
The last byte in this set, is the number of headers present in the SIP
message. After this byte, goes a section, called the Message Headers
Index, which gives quick access to each of the headers and their
codifications present in the message.
2.2.12.2. The headers index section
As it has been already discussed, the aim of SEAS project is to achieve
as high a performance as possible. One of the techniques enabling high
performance in text-based servers is the so called lazy parsing. To
enable the laziest possible parsing at the Application Server endpoint,
a mechanism has been used so that access to a requested SIP header can
be delayed until the application requests it, and the access can be
direct to that header, without parsing the former headers present in
the SIP message. Recall that one of the performance drawbacks of the
SIP protocol is that headers of any type can be spread all along the
header section, not having the constraint of putting the most critical
sip-specific headers at the beginning and ordered (which would be, in
fact, very desirable).
For this to be possible, there is a section right after the beginning
of the payload (the general message information section) which is a
kind of hash table, giving quick access to the codes (as explained in
the previous sections) of each of the headers present in the message.
This sort of hash table, is composed of triplets of bytes. The first
byte of each three is a code indicating which kind of header it points
to (whether it is a From, To, Call-ID, Route header, etc). Then follows
a 2 byte network-byte-order integer that points to a section in the
codified-header where the body of this header is more specifically
described.
This gives really fast access to any of the headers. For example, if
all the Route Headers were requested by the application, then a lookup
in this table would be necessary, looking for the value '9'
(corresponding to the Route header) in each of the positions multiple
of 3 (0,3,6,9,12, etc). This can be done in a extremely fast and easy
way, as this snipped of pseudo code explains:
for(int j=0,int i=0;i<table_length;i+=3){
if(payload[i]==9)
results[j++]=i;
}
this would let in the "results" array all the indexes in the headers
table that refer to a Route header. Then, the Route codification for
each of the headers could be reached thanks to the two-byte unsigned
integer that follows each of the header identifiers.
Figure 2.12. SEAS Headers Index section overview
SEAS Headers Index section overview
So a SIP message codified by the SEAS protocol, has the following
layout:
Figure 2.13. SEAS SIP-Message codification
SEAS SIP-Message codification
SIP Messages are a fundamental part of the protocol, but they are not
the only one. Transaction play a very important role in the SIP
protocol, within SER and in any JAIN-SIP implementation. For this
reason, the SEAS protocol also needs to define and implement some
semantics regarding transaction handling. The events related to a
transaction are: Incoming Request, Outgoing Request, Incoming Response,
Outgoing Response, Timeout and Transport Error.
So the SEAS protocol defines a specific format for each one of these
events. Internally, SER stores the transactions in a hash table. This
hash table generates an integer for each transaction applying a hash
function to its Via branch parameter, this integer is the hash index,
and it identifies in which slot within the hash table the transaction
is stored. The transaction table usually uses 65536 entries, so the
hash collision is pretty unlikely. Anyway, every hash entry is in
reality a linked list of transactions, so in the case a hash collision
(two transactions being assigned to the same hash slot) the
transactions are added to the same slot, each one being identified by
another integer called the label. The label within a hash slot, is
initially generated randomly, and then increased by one each time a
transaction falls in the same slot. So every transaction is identified
by a hash index and a label.
For incoming SIP requests, a transaction is generated at SER, and the
SEAS module gets that transaction identifier (hash index + label), then
grabs the source and destination IP, port and transport from every
message, and crafts a SEAS RequestIn event. This kind of event carries
all this information within it.
In order to send Responses out for the Server Transactions, JAIN can
send a type of Action messages, that order SER to send them to the
network. These messages follow a structure very similar to that of
RequestIn events: they start with the Action length in bytes, then
follows a byte giving the type of action, then follows the Hash Index
and the Label associated with the transaction that is being replied,
and finally the SIP Message in raw format. It doesn't use the SEAS
codification described above, because SER can easily parse the JAIN
provided Response to process it and send it out, so the pre-parsing is
not needed in that direction.
In order to generate Client Transactions, that is, sending SIP Requests
out, JAIN utilizes another kind of action called Seas Request Action.
In this case, when JAIN generates the Request to be sent out, it
doesn't have any means to know the transaction identifier (hash index
and label) that will be assigned to it by SER, so a new mechanism has
bee implemented to correlate JAIN requests to SER transactions.
Basically, JAIN-SIP assigns a unique identifier (an integer) that is
incremented by one for each new Client Transaction generated. This
identifier is passed to SER along with the SIP Request, so when a SIP
Response arrives to SER regarding that transaction, SER sends a
ResponseIn event to the JAIN stack, containing both the initial integer
identifying the transaction at JAIN and the hash index and label that
have been assigned to the transaction. This way, JAIN can correlate its
own identifiers with the identifiers used within SER.
Figure 2.14. Different kinds of SEAS codified Events and Actions
Different kinds of SEAS codified Events and Actions
In case there is a Transaction Timeout, it is notified to the JAIN SIP
Stack by passing it a Seas Incoming Response with a flag called Faked
Reply, and a Response code number 408 (Request Timeout).
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