SIP SIMPLE Client SDK

= Echo cancellation =

[[TOC(Design*, depth=1)]]

SIP SIMPLE client is a Python software library that allows for easy development of Internet communications end-points based on SIP and related protocols for voice, rich presence, session based instant messaging (IM), file transfers and desktop sharing. Other media types can be easily added by using an extensible high-level API. SIP SIMPLE client uses [http://www.pjsip.org/pjmedia/docs/html/index.htm PJSIP media library] for audio processing (sound card abstraction, audio codec and acoustic echo cancellation).

For more information see http://sipsimpleclient.com

The project is maintained by AG Projects that can provide help to the developer to integrate its AEC implementation.

== Background ==

The present implementation of the echo canceller from PJSIP media library is not performing to the desired level of quality. Practically, the speaker phone function of SIP SIMPLE client is not reliable and not performing satisfactory unless a headset is used.

[[Image(http://www.pjsip.org/images/media-flow.jpg)]]

== Project goal ==

Replace the existing AEC from Pjmedia library (what is displayed as echo.h in the above diagram whihc employes the AEC of the speex project) with an alternative to be developed solution that provides a high quality speakerphone user experience. The acoustic echo cancelation should perform comparable with any other proven VoIP commercial solution like Skype or iChat making the need of a headset un-necessary.

The developed software will be released under an open source licence and distributed with SIP SIMPLE client library.

The deliverable is C written program that can be applied to the pjsip cvs trunk 1.0 with the '''patch''' command.

== Resources ==

General

* Digital Signal Processing http://www.dsptutor.freeuk.com/
* Graduation thesis about AEC http://innovexpo.itee.uq.edu.au/2003/exhibits/s365914/
* DSP concepts: http://www.bores.com/courses/intro/basics/index.htm

PJSIP media library

* PJSIP media library http://pjsip.org, used by this project, AEC is implemented based on the Speex library, the code in pjsip/pjmedia/src/pjmedia/echo_speex.c
* Media flow - http://trac.pjsip.org/repos/wiki/media-flow
* PJSIP AEC API http://www.pjsip.org/pjmedia/docs/html/group__PJMEDIA__Echo__Cancel.htm

=== Speex AEC ===

This is the standard used AEC algorithm in PJSIP that needs to be replaced by a better solution

* Speex project http://speex.org, the echo canceller implements AEC based on the MDF algorithm, see speex/libspeex/mdf.c. A test client is available in speex/speexclient, this has been used for developing pjmedia equivalent
* http://speex.org/docs/api/speex-api-reference/group__SpeexEchoState.html

=== Andree Adrian AEC ===

* Draft AEC http://www.andreadrian.de/echo_cancel/draft-aec-03.txt
* http://www.andreadrian.de/intercom/ implements AEC based on NLMS algorithm (not tested yet)
* Kphone AEC implemented using above AEC http://www.andreadrian.de/echo_cancel/ (not tested)

This AEC has been implemented in C and integrated added as a working patch to the SIP SIMPLE project. It does not work properly, audio artifacts being present and further debugging requires DSP knowledge which the developer did not have. knowledge.

The actual implementation can be used as working example for how a third party AEC algorithm can be integrated with the SIP SIMPLE project.

== Next steps ==

There are two choices for going further:

1. Implement an AEC from scratch
2. Debug and fix Andree Adrian AEC