Sip Soft Switch Astri Con Adapted

7/29/2019 Sip Soft Switch Astri Con Adapted

1/28

Building a SIP softswitch withAsterisk and Asterisk-Java

Monica McArthur

Adapted from my presentation at AstriCon 2007


2/28

The task at hand

Build a pure SIP softswitch that can perform the following functions:

Answer an inbound call and redirect it to a specified target phonenumber

Rules for determining target number can be complex

Record both legs of the call

Play prerecorded prompts separately to inbound and outbound legs Provide call routing through IVR trees, overflow on busy/no answer,

and voicemail

Save the call detail record to a database for access by applications todo reporting and further data processing

All features can be provisioned in real-time

System must be highly-available and scalable Initial capacity 1250 simultaneous calls

99.99% availability requirement

Scheduled maintenance can be performed with no downtime


3/28

General solution

Write a routing application in Java to handle routing rules,provisioning changes, and interface with the database and otherapplications

Have the Java application direct a third-party host media processingsystem that provides the actual SIP signaling, RTP media handling,

prompt playing, call recording, and DTMF input

The application servers running the Java routing application are load-balanced by hardware and can be scaled as needed

The host media processing servers are load-balanced by a SIP proxy

server and can be scaled as needed


4/28

Particular issues

Inbound leg must be able to continue even if outbound leg fails To provide voicemail and overflow routing

Outbound leg must be able to continue after inbound leg hangs up ona connected call

To provide post-call input and play index number for recorded calls

Outbound calls may play a whisper heard only by the target while

the inbound still hears ringback

If there is no whisper, the call must support early media Media begins streaming (initially ringback) before outbound connect and must be

played to inbound

If early media is not streamed to inbound leg, the initial few syllables of the outboundcall may be lost (media clipping)

Ringback must not be included in call recordings


5/28

Overview of early media issue

In many standard SIP signaling exchanges, the answering user agentmay start generating media before the first agent is ready for it(media clipping)

To avoid media clipping, the answering user agent may send a 183(session progress) and then initiate a one-way media stream at thatpoint

If the initial user agent ignores the media stream sent with the 183and only accesses it after receiving a 200, media clipping will stilloccur

This issue is discussed in RFC 3960: Early Media and Ringing ToneGeneration in the Session Initiation Protocol (SIP)

http://www.rfc-archive.org/getrfc.php?rfc=3960
http://www.rfc-archive.org/getrfc.php?rfc=3960http://www.rfc-archive.org/getrfc.php?rfc=3960http://www.rfc-archive.org/getrfc.php?rfc=3960http://www.rfc-archive.org/getrfc.php?rfc=3960


6/28

Diagram of SIP signaling with early media

Media path established

Inbound leg Media gateway Outbound leg

INVITE

100 TRYING

183 SESSION PROGRESS

Early media

200 OK

Normal media

Media that can be lost

to clipping


7/28

Problems with existing solution

Ports are expensive!

Chosen host media platform did not support early media

Dependent on vendor to implement fixes Could take years

Often broke workarounds in place to support other features


8/28

New solution: Asterisk

Port cost now zero

More port density per server (can easily achieve 150 vs. 125)

Open source allows us to either find bug fixes in the Asteriskcommunity or write our own

FastAGI and AMI provide means for existing Java software to

communicate with Asterisk in ways similar to the previous HMP Only need to replace the vendor-specific code


9/28

Software architecture with Asterisk


10/28

How to interface with Asterisk

Could write our own software to interface with FastAGI and AMI

Or could select from a wide variety of existing open source libraries

After review, selected Asterisk-Java

http://asterisk-java.org
http://asterisk-java.org/http://asterisk-java.org/http://asterisk-java.org/http://asterisk-java.org/


11/28

Asterisk-Java

Open source, free library for Asterisk integration

Hosted in SourceForge

Current version is 0.3

Handles the low-level details of FastAGI and AMI communication

Java code for accessing AGI using Asterisk-Java is structuredsimilarly to servlets

AMI communication is handled through ManagerActions (to send AMIactions) and ManagerEvents (to receive AMI events)


12/28

Accessing AGI in Asterisk-Java

AGI applications are implemented as subclasses of BaseAgiScript

BaseAgiScript provides convenience methods to send all AGIcommands

AGI scripts are mapped to correct classes in setup code

service() method of BaseAgiScript has two arguments, AgiRequestand AgiChannel

AgiRequest contains information about the call (caller ID, dialeddigits, etc.)

AgiChannel handles the details of the convenience methods


13/28

Accessing AGI in Asterisk-Java(examples)

public class AGIInbound extends BaseAgiScript {

public void service(AgiRequest request, AgiChannel channel)

throws AgiException {

callID = new Long(getVariable(APP_CALLID"));this.streamFile(greetingFile);

this.streamFile("routing/tone");

this.exec("Record", recordingFilename + ".wav" + "|" + silence

+ "|" + maxduration + "|q");

this.hangup();

Agi script to begin handling call

Code for voicemail

agiMap.put("AGIInbound.agi", new AGIInbound());

SimpleMappingStrategy agiMapping = new SimpleMappingStrategy();

agiMapping.setMappings(agiMap);

agiServer.setMappingStrategy(agiMapping);

Setting up the mapping for AGIInbound


14/28

Accessing AMI in Asterisk-Java

Subclasses of ManagerAction are provided for each AMI action e.g., OriginateAction, HangupAction, SetVarAction

Instances of actions are sent by using an instance ofManagerConnection

Subclasses of ManagerEvent are provided for each AMI event e.g., DialEvent, HangupEvent, NewChannelEvent

Subclasses of ManagerEventListener are registered to listen on aManagerConnection

Can also create custom events that are subclasses of ManagerEventand register them with the ManagerConnection


15/28

Accessing AMI in Asterisk-Java(examples)

managerConnection.addEventListener(new

ManagerEventListenerProxy(amiDispatchers.get(routingNode)));

managerConnection.registerUserEventClass(Class.forName(

"astrouting.control.ami.events.ConnectedEvent"));

public class ConnectedEvent extends ManagerEvent {

private String channelName;

private String channelID;

private String userData;

Configure event listener and custom event

Custom ManagerEvent

OriginateAction originateAction = new OriginateAction();originateAction.setChannel(channel);

originateAction.setVariable(APP_CALLID", "" + astCall.getCallID());

originateAction.setAsync(true);

originateAction.setTimeout(1000L*astCall.getCall().getRNATime()+5000L);

managerResponse = managerConnection.sendAction(originateAction);

Create and send OriginateAction


16/28

Software architecture (detailed)


17/28

Remaining issues

With this architecture and a plain version of Asterisk, we can provide allof the required features of the softswitch EXCEPT

Having the outbound leg survive after the inbound disconnects Need legs in separate threads

Early media app_dial does provide support for early media, but only with the channel it is running

on


18/28

Problem 1: having the outbound legsurvive after the inbound disconnects

The straightforward way to handle connecting an inbound leg toanother number is to dial the number using app_dial

Unfortunately, in that case the outbound leg does not survive thehangup of the inbound leg

Need to have each leg living independently (in its own channel) butstill joined together


19/28

Solution

To get the outbound leg in its own channel: use AMI Originate to get alocal channel, connect to AGI, then use app_dial to make theoutbound call

To join the two channels together: use a patch for bridgingindependent legs

Was bug 5841; in trunk for 1.6


20/28

Originate on a local channel

First, use AMI OriginateAction to request a local channel that will startin a particular context and go to another context when connected

Syntax is local/s@

Then, launch AGI script from context

In AGI script, use app_dial to make actual call for outbound leg

Check result of app_dial in start context to handle busy/no answer

Perform additional functions on connected leg in context specified forconnection

This is a well-known pattern; see

http://blogs.reucon.com/asterisk-java/2007/04/18/originate_using_asterisk_local_channels.html
http://blogs.reucon.com/asterisk-java/2007/04/18/originate_using_asterisk_local_channels.htmlhttp://blogs.reucon.com/asterisk-java/2007/04/18/originate_using_asterisk_local_channels.htmlhttp://blogs.reucon.com/asterisk-java/2007/04/18/originate_using_asterisk_local_channels.htmlhttp://blogs.reucon.com/asterisk-java/2007/04/18/originate_using_asterisk_local_channels.htmlhttp://blogs.reucon.com/asterisk-java/2007/04/18/originate_using_asterisk_local_channels.html


21/28

Example code for origination

int dialExecResult = exec(Dial, "SIP/" + target + "@nextone|" +

dialTimeout);

String DIALSTATUS = this.getVariable("DIALSTATUS");

if (dialExecResult == 0) {

if ("NOANSWER".equals(DIALSTATUS)) {

astCall.noAnswer();

} else if ("BUSY".equals(DIALSTATUS)) {

astCall.busy();}

// etc. for CONGESTION, CHANUNAVAIL, CANCEL, HANGUP, default

}

Configure and send OriginateActionOriginateAction originateAction = new OriginateAction();

originateAction.setChannel("Local/s@ob-agi-dial");

originateAction.setApplication("Agi");

originateAction.setData("agi://" + agiServer+

":4573/AGIOutboundConnect.agi");

managerResponse = managerConnection.sendAction(originateAction);

Launch app_dial and check result


22/28

Bridge patch

Bridge patch was originally submitted with bug 5841: Bridge twochannels via a Dialplan App or an AMI event

Provides a Bridge() application for dialplan/AGI and an AMI Bridgeaction that will bridge the current channel with another specifiedchannel that already exists

Used to bridge the inbound leg with the connected outbound legobtained by app_dial

Patch we used was bridge-trunk-rev48286.patch

Code is now included in 1.6 trunk See http://bugs.digium.com/view.php?id=5841 for details
http://bugs.digium.com/view.php?id=5841http://bugs.digium.com/view.php?id=5841


23/28

Problem 2: early media

app_dial provides support for early media, but only to its inbound leg

In this architecture, the inbound leg is a local channel and the actualinbound leg does not receive media from the outbound leg until theyare joined using Bridge()

In order to provide early media to the inbound leg, app_dial needs toreturn if SIP 183 (session progress) is received

Since calls with whisper cannot use early media, whether app_dialreturns on SIP 183 needs to be configurable

In order to avoid recording the ringback when early media is used, theJava application needs to know when the SIP 200 (OK) is receivedafter app_dial connects on SIP 183


24/28

Example code to use bridge patch

public void onManagerEvent(ManagerEvent event) {

else if (event instanceof DialEvent) {

astCall = AstRoutingController.getController().

getAstCallByChannel(event.getSrc());

astCall.setObAstChannel(event.getDestination());

}

Get outbound channel ID from DialEvent

exec("Bridge", astCall.getObAstChannel());

Execute bridge


25/28

Solution: app_dial and channel patch

This required an original patch

Not yet submitted to Asterisk; will consider based on demand

app_dial.c changed to have new argument which specifies whether toconnect on SIP 183 if received

app_dial.c also stores which signal (PROGRESS/183 or ANSWER/200)it actually connected on SIP spec does not require answering user agent to send 183

channel.c changed to send custom AMI event on receipt of answer Used to determine time to start recording if app_dial connected on 183


26/28

Example code to use the new patch

String dialExecString = "SIP/" + target + "@nextone|" + dialTimeout;

if(astCall.earlyMedia())

dialExecString += ||1";

int dialExecResult = exec(Dial, dialExecString);

public void onManagerEvent(ManagerEvent event) {

else if (event instanceof ConnectedEvent) {

astCall = AstRoutingController.getController().

getAstCallByChannel(event.getChannelName());

astCall.connectSignaled();

}

Call app_dial with new argument

Receive notice of connect

String connectedSignal = getVariable("CONNECTED_SIGNAL");if("PROGRESS".equals(connectedSignal)) {

// handle early media

} else {

// handle normal flow

}

Check channel variable


27/28


28/28

Summary of best practices learned

Go into an AGI script in a context immediately

Use AMI events (hangup events, dial events, new events as needed)to keep track of call state and handle graceful hangups

To get an outbound leg in its own thread, originate on a local channel

and then use app_dial (called from an AGI script) to make the actualoutbound call

Sip Soft Switch Astri Con Adapted

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