[asterisk-dev] [Code Review] SIP project planning page

[Mark Michelson]

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https://reviewboard.asterisk.org/r/2183/
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Review request for Asterisk Developers.


Summary
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Have a look at the SIP project page that has been created here: https://wiki.asterisk.org/wiki/display/AST/New+SIP+channel+driver as well as its subpages.

At the moment, the content is pretty bare. This is because the project is not very far underway and so there isn't much to be able to put there right now.

A few notes:

The list of features on the project page is a list that more or less came off the top of my head and is supposed to represent a high level set of features. Feel free to let me know about things that should be added, but keep in mind that features listed there should be SIP-specific. We are not planning to do many changes to the media handling in Asterisk as of right now, so features like SRTP, STUN/TURN/ICE, and DTLS-SRTP are already provided by Asterisk's media handling and does not need to be addressed as a SIP feature (aside from potentially some SDP parsing).

The use cases are written from a *very* high level. For instance, the two-party call use case has dozens, if not hundreds, of variants. However, since these are not necessarily observable by Alice and Bob, they have not been included as separate use cases. So when evaluating use cases, information such as codecs in use, encryption of media, authentication, direct media, and NAT are irrelevant. Keep this in mind if you have additional use cases for me to document.

Please let me know what all can be added, removed, or clarified.


Diffs
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Diff: https://reviewboard.asterisk.org/r/2183/diff


Testing
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Thanks,

Mark

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