Age | Commit message (Collapse) | Author |
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- Fixed pjsua_media_channel_create_sdp() to re-calculate audio index of the remote offer, instead of using existing audio index calculated by pjsua_media_channel_init(), as for subsequent SDP offer/answer, pjsua_media_channel_init() may not be called.
- Fixed SRTP transport to be able to switch SRTP status from active to inactive/by-passed and vice versa.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3376 74dad513-b988-da41-8d7b-12977e46ad98
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lost between two requests (thanks Nikolay Popok for the report)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3375 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed lock codec to always be done after successful media update, and pend the lock codec until call state CONFIRMED if media update is done in call state EARLY but remote does not support UPDATE method.
- Added additional checks in lock_codec() and perform_lock_codec(), e.g: skip locking codec when media deactivated.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3374 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3371 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3368 74dad513-b988-da41-8d7b-12977e46ad98
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Contact when re-registering if the server does not support SIP outbound
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3367 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3366 74dad513-b988-da41-8d7b-12977e46ad98
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Johan Lantz for the suggestion)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3363 74dad513-b988-da41-8d7b-12977e46ad98
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requests (thanks Marcus Froeschl for the suggestion)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3361 74dad513-b988-da41-8d7b-12977e46ad98
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multiple codecs (thanks Cyril GY for the report)):
- avoid using pre-created SDP, but rather use timer and create SDP right when the UPDATE/re-INVITE is about to be sent, to avoid the use of stale pool
- also fixed bug in the old code when the lock codec feature is not activated after the call is confirmed
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3349 74dad513-b988-da41-8d7b-12977e46ad98
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the report)): leave the quote in parameter values and let the multipart code handle this instead
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3342 74dad513-b988-da41-8d7b-12977e46ad98
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NULL event parameter as application may not expect this
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3341 74dad513-b988-da41-8d7b-12977e46ad98
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the report)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3340 74dad513-b988-da41-8d7b-12977e46ad98
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Johan Lantz for the suggestion):
- added on_buddy_evsub_state() callback
- added sample implementation in pjsua_app.c
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3339 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3337 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed no audio bug when pjsua with SRTP optional-with-duplicated-offer calls pjsua with SRTP disabled, by updating active media index after SDP negotiation done.
- Fixed bug in generating SDP, pjsua_media_channel_create_sdp(), by making sure all media in the SDP candidate are aligned with current active SDP before calling pjmedia_transport_encode_sdp().
- Fixed bug in modifying SDP for call hold, the media index to be modified was hardcoded to 0, should be active media index.
- Added python tests for calls with SRTP optional-with-duplicated-offer.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3334 74dad513-b988-da41-8d7b-12977e46ad98
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Although RFC 3265 says it is only optional, some downstream RFC may bring this requirement to SHOULD strength - e.g. RFC 5373 (thanks Johan Lantz for the suggestion)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3331 74dad513-b988-da41-8d7b-12977e46ad98
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should be used when putting call on hold):
- use PJSUA_CALL_HOLD_TYPE_DEFAULT to specify default global call hold type
- use pjsua_acc_config.call_hold_type to specify call hold type for the account
- call hold type can also be set on per call basis by changing the call_hold_type in the call structure (requires inclusion of <pjsua-lib/pjsua_internal.h>
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3330 74dad513-b988-da41-8d7b-12977e46ad98
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in URI in To and From header (thanks Marcus Froeschl for the suggestion)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3329 74dad513-b988-da41-8d7b-12977e46ad98
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race condition when setting up transaction timeout
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3328 74dad513-b988-da41-8d7b-12977e46ad98
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custom headers for REGISTER requests of the account.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3326 74dad513-b988-da41-8d7b-12977e46ad98
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received (thanks Montevecchi Massimiliano and Klaus Darilion for the report)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3324 74dad513-b988-da41-8d7b-12977e46ad98
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- added new PJSUA API: pjsua_verify_url() which can be used for tel: URI
- modified and tested according to spec
- added new PJSIP error code, PJSIP_ENOROUTESET, to indicate that route set is needed to send to tel: URI
- added couple of unit tests (we can't cover the whole tel: URI scenario yet)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3323 74dad513-b988-da41-8d7b-12977e46ad98
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- Added new pjsua registration status callback on_reg_state2(), it includes the whole info from the lower layer registration callback pjsip_regc_cb().
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3322 74dad513-b988-da41-8d7b-12977e46ad98
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documentation when WWWDIR is specified
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3319 74dad513-b988-da41-8d7b-12977e46ad98
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- Added run-time configuration for activating/deactivating stream keep-alive (non-codec-VAD mechanism), also added this config to account settings.
- Fixed bug wrong session info pointer "si" in pjsua_media_channel_update() when call audio index is not zero.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3313 74dad513-b988-da41-8d7b-12977e46ad98
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Struble for the report))
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3311 74dad513-b988-da41-8d7b-12977e46ad98
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- Added enum pjsua_sip_timer_use for session timer usage types, containing: inactive, optional, required, always
- Replaced require_timer (boolean) with above enum in global and account config setting.
- Updated pjsua app --use-timer option accordingly.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3305 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3304 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3303 74dad513-b988-da41-8d7b-12977e46ad98
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because network connectivity is lost (thanks Robbie Hanson for the fix!)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3302 74dad513-b988-da41-8d7b-12977e46ad98
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request contains escaped characters (thanks Ferguen Adel for the report)):
- Fixed the printing part of Via "branch" parameter and To/From "tag" parameter, since these parameters are important for transaction/dialog identification
- Note that if the escaping sequence describes a character that otherwise is a valid token, that token would still be printed unescaped, hence the problem would still persist. But sender really shouldn't send this kind of escaped sequence as it really is asking for trouble.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3301 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3274 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3273 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3272 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3271 74dad513-b988-da41-8d7b-12977e46ad98
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Windows Mobile build configs (obsoleted by sip_transport_tls.c).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3264 74dad513-b988-da41-8d7b-12977e46ad98
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2005
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3262 74dad513-b988-da41-8d7b-12977e46ad98
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already has one (thanks Rafael Maia for the suggestion)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3260 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3255 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3244 74dad513-b988-da41-8d7b-12977e46ad98
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- incoming multipart message will be handled automatically
- for testing, enable HAVE_MULTIPART_TEST in pjsua_app.c
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3243 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3242 74dad513-b988-da41-8d7b-12977e46ad98
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pjsip_media_type from a simple string to pjsip_param, to support a more complex use of this field
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3241 74dad513-b988-da41-8d7b-12977e46ad98
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- Added (back) raw jitter statistics into RTCP statistics, with the new name "rx_raw_jitter".
- Added IPDV statistics into RTCP statistics.
- Added new compile-time settings to enable/disable raw jitter and IPDV statistics.
- Updated call dump in pjsua-lib.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3239 74dad513-b988-da41-8d7b-12977e46ad98
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compiler decides to use 16bit integer to represent this enum. In PJSUA-LIB, there is a code which assigns 32bit value to a variable of this type, causing overflow. Thanks Rickard Angbratt for the report
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3233 74dad513-b988-da41-8d7b-12977e46ad98
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Anens for the info)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3223 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3222 74dad513-b988-da41-8d7b-12977e46ad98
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scenario). Details:
- now the stream will be destroyed but the media transport will be kept alive during doublehold scenario
- small fix in SRTP to also negotiate crypto even when the media is marked as inactive, otherwise it's possible that an "optional" endpoint would create RTP/AVP offer and send it to "mandatory" endpoint, which would be rejected and cause the media port to be set to zero
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3219 74dad513-b988-da41-8d7b-12977e46ad98
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- fixed unterminated negotiation if our media transport rejects incoming offer (e.g. due to mismatch SRTP transport) with 488.
- to fix the above, modified the SDP negotiator (sdp_neg.[hc])'s pjmedia_sdp_neg_cancel_offer() to also be able to cancel in remote offer state
- also fixed the bug introduced previous Session Timer fix (Re: #1047), which cause SDP negotiator's state to be cleared after failed UAC UPDATE transaction is terminated, which means UPDATE can only be sent 5 seconds after the last UPDATE if the last UPDATE failed.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3217 74dad513-b988-da41-8d7b-12977e46ad98
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