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disconnected
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3771 74dad513-b988-da41-8d7b-12977e46ad98
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account config with pjsua_acc_modify()!
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3770 74dad513-b988-da41-8d7b-12977e46ad98
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registration retry
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3769 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3768 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3762 74dad513-b988-da41-8d7b-12977e46ad98
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inside a callback. The following APIs should be safe to be called in this situation:
- pjsua_call_get_info()
- pjsua_call_get_conf_port()
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3751 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3749 74dad513-b988-da41-8d7b-12977e46ad98
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if the display part of the AOR contains reserved characters. This fixes #1363
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3742 74dad513-b988-da41-8d7b-12977e46ad98
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accounts for presence timer update. Thanks Matt DiMeo for the patch
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3604 74dad513-b988-da41-8d7b-12977e46ad98
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and closed #1314 (New callback to notify application when registration or unregistration has been initiated). Thanks Tony Jago Million for the patch
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3594 74dad513-b988-da41-8d7b-12977e46ad98
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- protected the account validation in the registration callback.
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3591 74dad513-b988-da41-8d7b-12977e46ad98
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application
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3585 74dad513-b988-da41-8d7b-12977e46ad98
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van der Bent for the report)
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3584 74dad513-b988-da41-8d7b-12977e46ad98
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indefinitely (thanks Attila Magyar for the fix)
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3576 74dad513-b988-da41-8d7b-12977e46ad98
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- fixed bug in SIP invite module, SDP negotiator state should be reverted back after an SDP re-offer is rejected by application.
- fixed bug in pjsua_call_on_rx_offer(), evaluating call->audio_index should be done after pjsua_media_channel_create_sdp() is successful.
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3574 74dad513-b988-da41-8d7b-12977e46ad98
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stream.
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3571 74dad513-b988-da41-8d7b-12977e46ad98
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Marcus Froeschl for the suggestion)
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3570 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3566 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3553 74dad513-b988-da41-8d7b-12977e46ad98
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(thanks Viktor Krikun for the report)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3538 74dad513-b988-da41-8d7b-12977e46ad98
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counter in updating credential info.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3490 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3474 74dad513-b988-da41-8d7b-12977e46ad98
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re-INVITE or UPDATE
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3452 74dad513-b988-da41-8d7b-12977e46ad98
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rescheduled with the new delay. This can be useful if app wants to correct the delay after it checks the timeout in the callback.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3444 74dad513-b988-da41-8d7b-12977e46ad98
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pjsua_acc_config
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3441 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3438 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3429 74dad513-b988-da41-8d7b-12977e46ad98
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Linux/Unix
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3423 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3412 74dad513-b988-da41-8d7b-12977e46ad98
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removed):
- pjsua_media.c checks if audio media is present in the offer; if not, do not set any answer
- sip_inv.c checks if app has supplied an answer after on_rx_offer() callback is called, and returnd 488 (Not Acceptable) if not (previously, it will return 200/OK without SDP!)
- added a SIPp scenario file to reproduce this
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3383 74dad513-b988-da41-8d7b-12977e46ad98
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contact_params, contact_uri_params, and auth_pref, were not duplicated properly (thanks Roman Grachev for the report and the patch).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3377 74dad513-b988-da41-8d7b-12977e46ad98
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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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- 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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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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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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- 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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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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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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- 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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- 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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- 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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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3255 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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