[asterisk-dev] GSoC 2010: Testing the SDP parser in chan_sip

[Loren Sands-Ramshaw]

Hello all,

I'm wondering whether I'm on the right track in my general thinking?

- Look at process_sdp and the functions it calls to find unit test
candidates, and refactor in order to create unit test candidates where
appropriate.
- Create tests that give input to these functions and then verify that
the functions executed correctly - not only checking that the state is
set correctly (looks like mainly sip_pvt) but also that things are set
up correctly (eg- both that the ast_rtp_instance has good values and
that the port was opened). Also check correct return values and
logging in the case of bad input.
- Write code to generate session descriptions with varying errors and
varying combinations of session, time, and media type/value pairs,
exercising the range of possible configurations listed in SDP's RFC.

Also, where would you recommend I gather real-world examples of
session descriptions from? Are there common deviations from the RFC
that Asterisk supports or wants to support?

Thank you,
Loren