SRTP development

[nanang]

In r1697 imported libsrtp library to third_party directory.

[nanang]

In r1698:

• Add directory srtp to third_party/build directory
• Add libsrtp project & integrate it to pjproject vs8 solution
• Add transport_srtp.h & .c
• Modify project dependencies to include libsrtp
• Modify Samples-vc.mak, add libsrtp as third party library
• Modify transport interface
• Modify transport_ice & transport_udp to accomodate new transport interface
• Modify other files that uses transport

[bennylp]

In r1699:

• Added Visual Studion 6 project
• Changed config.h to compile in VS6
• Removed invalid reference to config files in VS8 project

[nanang]

In r1700:

• change libsrtp on rand_source.c to disable rand_s which causing breakpoint assert
• change pjsua_media.c:
  • allow RTP/SAVP transport
  • set srtp transport creation option to 0 (prev:auto destroy underlying transport is enabled)
• new pjmedia errno:

PJMEDIA_SRTP_ESDPREQSECTP: Secure transport required in SDP media descriptor.
PJMEDIA_SRTP_ESDPAMBIGUEANS: SDP contains ambigue answer.

• transport_srtp:
  • updating log & error code
  • bug fix: release pool after destroying other objects

[nanang]

In r1701:
put a note about disabling the 'rand_s' block in libsrtp file rand_source.c:111,
it causes assertion on vs8.

[bennylp]

In r1702:

• changed SRTP API according to test document specification
• added media_create() callback with media_index.

[bennylp]

r1707:

• Added PJSIP_ESESSIONINSECURE error code in sip_errno to require that secure session shall be used

r1708:

• updated ICE media transport's media_create() to consider the media index when updating SDP

r1709:

• SRTP transport is now created and destroyed on every call (ie. it's not reused)
• Added use_srtp option in pjsua_account_config to control SRTP usage (e.g. disabled, optional, or mandatory)
• Added srtp_secure_signaling option in pjsua_account_config to control whether SRTP requires secure signaling

[nanang]

r1710:
Updated errno.h & .c to accomodate SRTP Transport & libsrtp error code

[nanang]

r1711:

• implemented new flow based on new SRTP transport interface and data structure
• applied 3 SRTP operation mode: DISABLED, OPTIONAL, MANDATORY
• applied new error codes

[nanang]

r1712: changed libsrtp error string resolver

[nanang]

r1713:
changed transport_srtp.c: (re)applied setting close member transport & added more validation checks

[bennylp]

In r1718:

• Added and updated Makefiles for SRTP (tested on OSX)
• SRTP's config.h renamed to srtp_config.h since it conflicts with speex's config.h
• Updated few libsrtp files to use srtp_config.h instead
• Fixed const warnings in pjsua_call.c
• Fixed compilation error in streamutil.c
• Fixed compilation warning about duplicate declaration of pjmedia_sdp_session in pjmedia/transport.h

[bennylp]

In r1719:

• Updated VS6/VS8 project to use srtp_config.h
• Added srtp_err.c to integrate libsrtp logging to pjlib logging
• Updated VS6/VS8/Makefile files for above
• Updated PJSUA-LIB to always instantiate SRTP transport
• Fixed bug in pjsua_app.c with regard to --use-srtp option handling

[nanang]

r1720:
Added fields use_srtp and srtp_secure_signaling to pjsua_config as lookup fields for pjsua_acc_config_default.

[nanang]

r1721:
bugs fixed in transport_srtp.c:

• default crypto-suites in pjmedia_srtp_setting_default without crypto 'NULL'
• problem in bypass SRTP mechanism

[nanang]

In r1722:

• Bug fixed in mandatory mode, change local media sdp transport from RTP/AVP to RTP/SAVP
• Bug fixed for setting libsrtp flags (no encrypt, no auth)
• Changed log level & messages
• Put some more comments

[nanang]

In r1723:

• Applied preprocessor PJMEDIA_HAS_SRTP in PJMEDIA & PJSUA.
• Applied preprocessor PJSUA_DEFAULT_USE_SRTP and PJSUA_DEFAULT_SRTP_SECURE_SIGNALING in PJSUA.

[nanang]

r1724: Bug fixed in remote crypto variable to be compared to local offers in media_start.

[nanang]

In r1726:

• added part to test specific key, only for testing purpose
• replaced pool instance for parsing SDP and generating key (from suplied pool param of media_create/start to internal srtp pool)
• bug fixed in send rtcp callback
• changed mutex type from null mutex to recursive mutex
• changed srtp transport destroy: using mutex and reordering step.

[bennylp]

Moved to TRUNK!

[bennylp]

See ticket #61 for integration of this branch to trunk.