VoIP Asterisk&OpenSIPS Architecture

What’s SIP
• IETF RFC 3261

– “The Session Initiation Protocol (SIP) is an application-
layer control (signaling) protocol for creating, modifying
and terminating sessions with one or more
participants.”
• Can be used for voice, video, instant messaging, gaming,
etc., etc., etc.
• Follows on HTTP
– Text based messaging
– URIs – ex: sip:dbaron@MIT.EDU
Where’s SIP
SDP codecs
Application RTSP SIP RTP DNS(SRV)
Transport TCP UDP
Network IP
Physical/Data Link Ethernet

SIP Components
• User Agents
– Clients – Make requests
– Servers – Accept requests
• Server types
– Redirect Server
– Proxy Server
– Registrar Server
– Location Server
• Gateways
SIP Methods
• INVITE Requests a session
• ACK Final response to the INVITE
• OPTIONS Ask for server capabilities
• CANCEL Cancels a pending request
• BYE Terminates a session
• REGISTER Sends user’s address to server

SIP Responses
• 1XX Provisional 100 Trying
• 2XX Successful 200 OK
• 3XX Redirection 302 Moved Temporarily
• 4XX Client Error 404 Not Found
• 5XX Server Error 504 Server Time-out
• 6XX Global Failure 603 Decline

SIP Flows - Registration
User Registrar Location
B MIT.EDU MIT.EDU
REGISTER: sip:dbaron@MIT.EDU
401 - Unauthorized
REGISTER: (add credentials)

sip:dbaron@MIT.EDU
Contact 18.18.2.4
200 - OK
SIP Flows - Basic
User User
A B
“Calls”
INVITE: sip:18.18.2.4
18.18.2.4
180 - Ringing Rings
200 - OK Answers
ACK
Talking RTP Talking
Hangs up BYE
200 - OK
SIP Servers
● Registrar Sever
● takes registration requests from UA's, and maintains information
as to their location.
● Location Sever
● This server replaces the user's AoR (Address of Record) with the
contact address registered by user.
SIP Servers
● Proxy Server
● A SIP proxy receives SIP messages (both requests and
responses), that are generated by UA's, and forwards them on
(either to the destination UA, or the next SIP Server in the route,
as specified in the SIP message itself).
● Redirect Server
● This conceptual SIP network entity receives requests from a proxy
or UA, and responds with a specific SIP response method, telling
the originator of the request that the user is not available at that
location, and indicates where the request should be sent in order
to try again.
SIP Servers
● Application Sever
● This is the place were different features are implemented, adding
value to the VoIP platform. Among common features you can find
advanced call control applications like call blocking or call baring,
prepaid system, interactive call routing.
● Media Servers
● In order to have an active control of what happens during a call, a
device has to be 'in the media stream', as it is often called...
SIP SERVERs(decisions)
SIP Servers have a wide array of possible ways to
handle incoming messages
● Where to route the request
● Proxy/Redirect/Reject
● Stateful/Stateless forwarding
● Record-routing
● Forking—parallel/sequential/none
● Authentication
● Loop detection
SIP SERVERs(decision factors)
These decisions can be made based on many
factors, such as:
● Message destination address (Request-URI)
● Type of request (method)
● Other message fields: From, To, Date, Priority
● Response to a Request
Proxy Servers
The SIP standard defines SIP proxies as
“elements that route SIP requests to user agent
servers (UAS) and SIP responses to user agent
clients (UAC).
● Stateful proxy
● Stateless proxy
Stateful Proxy(TM)
When stateful, the proxy processes transactions
rather than individual messages
A stateful proxy is aware of the state of transactions and message
history, and can therefore perform better-informed processing on
incoming messages
● Memory consumption
● Throughput
● Implementation complexity
● Underlying SIP Stack complexity
STATELESS PROXY
A stateless proxy is a “simple message forwarder”
● A stateless proxy cannot associate responses
● A stateless proxy cannot associate retransmissions
● If the message is lost, the proxy will not retransmit it.
Note: Stateless proxies need to use a routing algorithm that always

routes copies of the same message to the same destination
Proxy Forking
A forking proxy, in the context of a

communications system using Session Initiation
Protocol (SIP), is a server that attempts to find a
correspondent by searching several locations.
SIP proxy severs that route messages to more than one

destination are called forking proxies.
Clustering scenarios
O
Inbound- service LB
Fronting small/medium size PBX-based services for

scaling, LB and HA.
Incoming Yate /
calls
Asterisk /
Freeswitch
OpenSIPS boxes
Load Balancer
O
Inbound- Call Center
Controller for a Call Center

scaling, LB and HA. Standard
operator
Incoming
calls Business
operator
OpenSIPS
Load Balancer
Advanced
operator
O
May 4-5 OpenSIPS – clustering and balancing media servers , Amoocon 2009 Rostock,Germany
OpenSER
OpenSER
from an asterisk point of view

What is OpenSER?
OpenSER is:
OpenSER is:
- a SIP proxy
OpenSER is:
- a SIP proxy
- a SIP registrar
OpenSER is:
- a SIP proxy
- a SIP registrar
- a SIP router
OpenSER is:
- a SIP proxy
- a SIP registrar
- a SIP router
- a SIP redirect server
OpenSER is:
- a SIP proxy
- a SIP registrar
- a SIP router
- a SIP redirect server
- a SIP application server
OpenSER is not:
OpenSER is not:
- a B2BUA (back to back user agent)

OpenSER is not:

- a media server
OpenSER is not:

- a media server
- a soft phone
OpenSER architecture:
two main logical components

core modules (plug-ins)

• parsers (SIP/SDP/conf)
• memory system
• memory system
• transport layer
• memory system
• transport layer
• database interface
• memory system
• transport layer
• synch/lock mechanism
• memory system
• transport layer
• management interface
• memory system
• transport layer
• script variables
• parsers (SIP/SDP/conf) • SIP protocol (tm)

• memory system
• transport layer
• parsers (SIP/SDP/conf) • SIP protocol (tm)

• memory system • features
• transport layer
module 1
module 2
core
module n
OpenSER configuration:
openser.cfg
openser.cfg (three blocks)

global options
global options
modules:
loading and parameters
global options
modules:
loading and parameters
routing
global options
global options
- listen interfaces
global options
- listen interfaces
# UDP support
listen=udp:10.10.10.10:5060
global options
- listen interfaces
# TCP support
listen=tcp:10.10.10.10:5070
global options
- listen interfaces
# TLS support (port 5061)

listen=tls:10.10.10.10
global options
- listen interfaces
- logging
global options
- listen interfaces
- logging
# logging level
debug=3
global options
- listen interfaces
- logging
# syslog enabled
log_stderr=no
global options
- listen interfaces
- logging
# syslog enabled
log_stderr=no
log_facility=LOG_LOCAL3
global options
- listen interfaces
- logging
# syslog enabled
log_stderr=no
log_name=”oser”
global options
- listen interfaces
- logging
- number of processes
global options
- listen interfaces
- logging
fork=yes
global options
- listen interfaces
- logging
fork=yes
children=6 # UDP listeners
global options
- listen interfaces
- logging
fork=yes
children=6 # UDP listeners
tcp_children=5 #TCP listeners
global options
- listen interfaces
- logging
- protocol control
global options
- listen interfaces
- logging
- protocol control
disable_tcp=yes
global options
- listen interfaces
- logging
- protocol control
disable_tcp=yes
disable_tls=yes
global options
# Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php

# for a explanation of possible statements, functions and parameters.
#
####### Global Parameters #########
debug=3
log_stderror=no
fork=yes
children=4
disable_tcp=yes
listen=udp:192.168.1.2:5060
modules (loading and parameters)

- path location
- path location
mpath=”/usr/lib/openser/modules”
- path location
- module loading
- path location
- module loading
mpath=”/usr/lib/openser/modules”
loadmodule “tm.so”
- path location
- module loading
loadmodule “/usr/lib/openser/modules/tm.so”
- path location
- module loading
- module parameters
- path location
- module loading
- module parameters
modparam(“tm”,“T1_timer”,700)
####### Modules Section ########
# set module path

mpath="/usr/local/lib/openser/modules/"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")
routing
routing
- request route: route[n] {...}

routing

- provisional reply route: onreply_route[m] {...}
routing

- failure reply route: failure_route[p] {...}
routing

- request branch route: branch_route[q] {...}
routing

- request branch route: branch_route[q] {...}
- error route: error_route {...}
routing
####### Routing Logic ########
route{
if (loose_route()) { # mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
t_relay();
exit;;
};
if (!uri==myself) { # mark routing logic in request
append_hf("P-hint: outbound\r\n");
t_relay();
exit;
};
if (uri==myself && is_method("INVITE")) {
lookup(“location”);
t_relay();
exit;
};
}
Read from the network
Message parsing
SIP request SIP reply
route on_reply_route/failure_route
Processing (core/module functions)
Send to network
SIP message call flow
OpenSER/asterisk integration
- registrar/dispatcher for asterisk farm
asterisk
asterisk
OpenSER
asterisk
location db
- SIP router for asterisk farm

high # of channels
carrier 1
medium cps rate
asterisk
high # of channels
OpenSER carrier 2
asterisk high cps rate
asterisk
low # of channels
low cps rate carrier 3
- SIP router for asterisk farm
carrier 1
international
asterisk
OpenSER carrier 2
asterisk national
asterisk
carrier 3
911

VoIP Asterisk&OpenSIPS Architecture

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VoIP Asterisk&OpenSIPS Architecture

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What’s SIP

• IETF RFC 3261

Application RTSP SIP RTP DNS(SRV)

Transport TCP UDP

Physical/Data Link Ethernet

• ACK Final response to the INVITE

• OPTIONS Ask for server capabilities

• CANCEL Cancels a pending request

• BYE Terminates a session

• REGISTER Sends user’s address to server

• 2XX Successful 200 OK

• 3XX Redirection 302 Moved Temporarily

• 4XX Client Error 404 Not Found

• 5XX Server Error 504 Server Time-out

• 6XX Global Failure 603 Decline

REGISTER: (add credentials)

180 - Ringing Rings

Talking RTP Talking

Note: Stateless proxies need to use a routing algorithm that always

A forking proxy, in the context of a

SIP proxy severs that route messages to more than one

Fronting small/medium size PBX-based services for

Controller for a Call Center

from an asterisk point of view

- a B2BUA (back to back user agent)

- a B2BUA (back to back user agent)

- a B2BUA (back to back user agent)

two main logical components

core modules (plug-ins)

core modules (plug-ins)

core modules (plug-ins)

core modules (plug-ins)

core modules (plug-ins)

core modules (plug-ins)

core modules (plug-ins)

core modules (plug-ins)

core modules (plug-ins)

• parsers (SIP/SDP/conf) • SIP protocol (tm)

core modules (plug-ins)

• parsers (SIP/SDP/conf) • SIP protocol (tm)

openser.cfg (three blocks)

# TLS support (port 5061)

# Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php

modules (loading and parameters)

modules (loading and parameters)

modules (loading and parameters)

modules (loading and parameters)

modules (loading and parameters)

modules (loading and parameters)

modules (loading and parameters)

modules (loading and parameters)

modules (loading and parameters)

####### Modules Section ########

# set module path

- request route: route[n] {...}

- request route: route[n] {...}

- request route: route[n] {...}

- request route: route[n] {...}

- request route: route[n] {...}

SIP request SIP reply

Processing (core/module functions)

- registrar/dispatcher for asterisk farm

- SIP router for asterisk farm