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ICE negotiation failed). [Re #1263]
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3732 74dad513-b988-da41-8d7b-12977e46ad98
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- Updated maximum video tee ports in pjsua video preview to (PJSUA_MAX_CALLS+1).
- Removed video tee maximum ports compile-time setting, MAX_DST_PORT_COUNT.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3725 74dad513-b988-da41-8d7b-12977e46ad98
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window. Also added pjsua_vid_preview_param_default()
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3724 74dad513-b988-da41-8d7b-12977e46ad98
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- Generating a deactivated pre-answer media by cloning remote media. There was a case that the media transport in the offer is bad/unrecognized, PJSUA still generated the preanswer with RTP/AVP.
- When generating answer, it should apply max media count (max_audio/video_cnt in account setting) after SDP negotiation instead of in the pjsua_media_channel_init()). This will require PJSUA to perform SDP re-negotiation when the SDP answer get changed.
- Fixed media priority/acceptibility sorting, e.g: media with RTP/SAVP transport still got acceptable score in SRTP disabled mode, this messed up the algorithm of applying max media count setting.
- Fixed SDP negotiator to skip format match in generating answer when the pre-answer provided is deactivated (port 0).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3714 74dad513-b988-da41-8d7b-12977e46ad98
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- Added PJSUA_CALL_VID_STRM_NO_OP to occupy value 0 for the enum
- Added pjsua_call_vid_strm_op_param_default() to initialize pjsua_call_vid_strm_op_param
- Renamed pjsua_call_get_transport_info() to pjsua_call_get_med_transport_info()
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3694 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3684 74dad513-b988-da41-8d7b-12977e46ad98
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fields: max_audio_cnt, max_video_cnt, vid_in_auto_show, vid_out_auto_transmit, vid_cap_dev, vid_rend_dev.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3683 74dad513-b988-da41-8d7b-12977e46ad98
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for the report
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3678 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3675 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3667 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed compile warnings on vs2005
- Fixed compile error when PJMEDIA_HAS_VIDEO set to 0 on vs2005
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3666 74dad513-b988-da41-8d7b-12977e46ad98
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"2.0-pre-alpha-svn".
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3664 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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- 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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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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Details:
- added new account config setting: reg_use_proxy. This contains bitmask values to indicate whether outbound proxies and account proxies are to be added in the REGISTER request. Default value is to add both.
- added new pjsua cmdline option to control this: --reg-use-proxy
- miscellaneous minor fixes in other pjsua cmdline arguments
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3216 74dad513-b988-da41-8d7b-12977e46ad98
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Krakowski for the report)) and fixes #1097 (Support sending UPDATE without SDP). Details:
- Session timer fixes:
- will look at remote capability in Allow header
- if UPDATE is supported, will send UPDATE without SDP first.
If this fails, will send UPDATE with SDP
- otherwise will send re-INVITE
- PJSUA-LIB will look at dialog's remote capability to determine
whether re-INVITE or UPDATE should be sent to change default
addresses after ICE negotiation.
- pjsip_inv_update() now allows NULL offer, in which case the
UPDATE will be sent without SDP.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3215 74dad513-b988-da41-8d7b-12977e46ad98
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request): added contact_rewrite_method account config to control this. Default is to use the new mechanism, i.e. the single REGISTER method.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3213 74dad513-b988-da41-8d7b-12977e46ad98
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