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Category: SIP Speaker

SIPSpeaker is a system that provides answering machine service. When the system receives a SIP call, it will establish a new session with user client and send voice data to the user. The user can configure a customized message using a web interface.
The administrator of the system can configure the interface and port number for the web interface; he/she can also configure SIP service interface, port number as well as user name, either using a configuration file or command line arguments. Done in 2010.

Transferring Voice Data

This is done using JMF (Java Media Framework). Voice data is read from a file using MediaLocator'. It is used as the data source for a Processor’. The Processor' specifies and converts the audio data to a certain format and then output the converted data into a DataSink’. The `DataSink’ then transfers the stream to its destination address and port.

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Multiple Sessions

A SessionManager' is used to manage all the sessions. The SessionManager’ keeps a list of sessions. When it receives data, it will first check which session the data belongs to. Then the data is given to the corresponding session. If it does not belong to any session in the list, a new session will be created and added to the session list.

The basic code looks like this:

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Session Establishment and Tearing Down

Each session has a ‘status’, it can be ‘new’, ‘establishing’, ‘cancelling’, ‘established’, ‘tearingdown’ and ‘destroyed’.

When a new session is created, its status is ‘new’.

When an ‘INVITE’ is received, it sends out an ‘OK’ message and change its status to ‘establishing’.

After receiving an ‘ACK’ message the status will be changed to ‘established’.

Then begins the transferring of voice data using RTP.

When the sending finishes the status will become ‘tearingdown’.

A ‘BYE’ message is also sent to the client.

The status becomes ‘destroyed’ after getting ‘OK’ from the client.

When a ‘CANCEL’ message is received, the status becomes ‘cancelling’.

Then it sends back ‘OK’ and ‘Request Terminated’ messages.

After received an ‘ACK’, the status becomes ‘destroyed’.

The thread is as follows:

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Generating Voice File

When the web server receives a POST' message from the web page, it will first check whether the message is valid. If so, freeTTS‘ is used to generate a voice file and the file is saved in the `wav’ directory with pre-configured file name.

The voice generator looks as follows:

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SIP Speaker Overview

SIPSpeaker is a system that provides answering machine service. When the system receives a SIP call, it will establish a new session with user client and send voice data to the user. The user can configure a customized message using a web interface.

The administrator of the system can configure the interface and port number for the web interface; he/she can also configure SIP service interface, port number as well as user name, either using a configuration file or command line arguments.

The system supports multiple session to be communicated at the same time. It uses SIP to manage sessions, and SDP as the handshake protocol to reach an agreement on media format transfered on wire.

JMF is used to support audio playback. And FreeTTS is used to convert message from text to audio format.

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