- Currently only Works with raw video and audio AVI files
- Added --play-avi and --auto-play-avi options in pjsua
- No A/V synchronization yet
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@4016 74dad513-b988-da41-8d7b-12977e46ad98
* add PJSUA_MEDIA_HAS_PJMEDIA macro
* move pjmedia specific implementation in pjsua_media.c and pjsua_call.c into pjsua_aud.c
* add pjsip-apps/src/third_party_media sample containing:
- alt_pjsua_aud.c
- alt_pjsua_vid.c
* moved pjmedia_vid_stream_info_from_sdp() into pjmedia/vid_stream_info.c
* moved pjmedia_stream_info_from_sdp() into pjmedia/stream_info.c
* misc: fixed mips_test.c if codecs are disabled
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3982 74dad513-b988-da41-8d7b-12977e46ad98
- Add and manage pool instance in default codec param in video codec framework.
- API change: pool param is removed from pjmedia_vid_codec_mgr_set_default_param().
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3956 74dad513-b988-da41-8d7b-12977e46ad98
Add an API pjsua_schedule_timer2() to allow application to schedule a callback function to be executed after a specified time interval. This enables app to post a delayed job which, in this case, allows the initialization of all media transport creations to finish first before we get the media transport creations result.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3938 74dad513-b988-da41-8d7b-12977e46ad98
- Updating port info of the switchboard master port (after reopening audio device) with the audio device param should care about PJMEDIA_AUD_DEV_CAP_EXT_FORMAT flag, i.e: only copy from audio device extended format info when the flag is set.
- Fixed switchboard to update the master port info shortcut in connecting ports, as master port info may get updated to match to the connecting ports.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3915 74dad513-b988-da41-8d7b-12977e46ad98
- for transferee (attended & unattended): via new PJSUA-LIB callback on_call_transfer_request2()
- for transfer destination (attended only): via new PJSUA-LIB callback on_call_replace_request2()
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3903 74dad513-b988-da41-8d7b-12977e46ad98
- moved the media count setting from account setting to call setting
- introduced pjsua_call_setting, to be used by pjsua_call_make_call() and some new APIs: pjsua_call_answer2(), pjsua_call_reinvite2(), pjsua_call_update2()
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3891 74dad513-b988-da41-8d7b-12977e46ad98
* Backport of r3557:r3832
TODO: ticket #1268 (Option for automatic/manual sending of RTCP SDES/BYE for the stream) for video stream.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3841 74dad513-b988-da41-8d7b-12977e46ad98
This should work for all codecs, audio & video. Can be disabled at compile-time
using PJMEDIA_SDP_NEG_REWRITE_ANSWER_PT macro setting.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3837 74dad513-b988-da41-8d7b-12977e46ad98
* Make sure that all media transports are already created and completed to fix the assertion when making call using ICE.
* Change the callback pjsua_med_tp_state_cb to return pj_status_t (instead of void)
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3796 74dad513-b988-da41-8d7b-12977e46ad98
- Fixed wrong place of video capture & render event subscription initialization, causing it getting reinitted while being subscribed.
- Moved capture/render event unsubscription to be after capture/render port stopped. Also restart the capturer (after being stopped for unsubsciption & stream detachment) only when the capturer is being used by other, e.g: stream or preview.
- Fixed error handling in pjsua_call_reinvite(), call pjsip_dlg_dec_lock() only if dlg is successfully acquired.
- Minor: added [un]subscribtion log to event.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3778 74dad513-b988-da41-8d7b-12977e46ad98
- changed encode(), packetize(), unpacketize(), and decode() to encode_begin(), encode_more(), and decode()
- codec has new "packing" setting
- updated stream, aviplay, codec-test, and stream-util due to above
- minor doxygen documentation fixes here and there
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3776 74dad513-b988-da41-8d7b-12977e46ad98
* Add feature that allows ICE media transport to be created asynchronously.
* Add new callback, e.g. on_call_media_transport_state(call_id, state_struct) to report media transport status.
* Handle outgoing calls while creating media transport asynchronously.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3763 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3725 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3714 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3694 74dad513-b988-da41-8d7b-12977e46ad98
- Fixed compile warnings on vs2005
- Fixed compile error when PJMEDIA_HAS_VIDEO set to 0 on vs2005
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3666 74dad513-b988-da41-8d7b-12977e46ad98
- Replaced video stream operation DISABLE into REMOVE.
- Replaced video stream operation ENABLE into CHANGEDIR.
- Added new param: media direction, used in operation ADD and CHANGEDIR.
- Updated video stream operation START_TRANSMIT to ignore capture device param (as changing capture device is handled by CHANGE_CAP_DEV operation).
git-svn-id: https://svn.pjsip.org/repos/pjproject/branches/projects/2.0-dev@3657 74dad513-b988-da41-8d7b-12977e46ad98
- Implemented media info/statistics APIs: stream info, stream statistic, and transport info.
- Implemented API of default video stream index in call, pjsua_call_get_vid_stream_idx().
git-svn-id: https://svn.pjsip.org/repos/pjproject/branches/projects/2.0-dev@3639 74dad513-b988-da41-8d7b-12977e46ad98
- Break down the operation type PJSUA_CALL_VID_STRM_MODIFY into PJSUA_CALL_VID_STRM_ENABLE, PJSUA_CALL_VID_STRM_DISABLE, PJSUA_CALL_VID_STRM_CHANGE_CAP_DEV.
- Implemented video stream re-enabling (PJSUA_CALL_VID_STRM_ENABLE).
git-svn-id: https://svn.pjsip.org/repos/pjproject/branches/projects/2.0-dev@3636 74dad513-b988-da41-8d7b-12977e46ad98
- Renamed API pjsua_call_set_vid_out() with pjsua_call_set_vid_strm().
- Implemented initial version of the function, features covered:
- add, remove video media stream during the call
- change which device to use during the call
- start/stop video stream transmission
git-svn-id: https://svn.pjsip.org/repos/pjproject/branches/projects/2.0-dev@3634 74dad513-b988-da41-8d7b-12977e46ad98
- API designed and reviewed (pjsua.h)
- Implemented these APIs and added to pjsua sample application:
- video device enums API
- video capture preview API
- refactoring in PJSUA-LIB:
- video stuffs go to pjsua_vid.c
- call dump goes to pjsua_dump.c
We're still missing:
- video call API implementation
- media info and statistic API implementation
git-svn-id: https://svn.pjsip.org/repos/pjproject/branches/projects/2.0-dev@3609 74dad513-b988-da41-8d7b-12977e46ad98
- Fixed handling remote re-offer, where SDP media line may be added or removed.
- Fixed bug in receiving remote offer (initial or subsequent), media channel create sdp must consider acc->cfg.max_audio_cnt setting.
- Fixed bug media transport is not closed after call disconnected.
- Fixed assertion in lock_codec after receiving initial answer but no acceptable media (in pjsua level, e.g: SRTP nego failed), now the call will be terminated.
git-svn-id: https://svn.pjsip.org/repos/pjproject/branches/projects/2.0-dev@3560 74dad513-b988-da41-8d7b-12977e46ad98
- Added custom negotiation callback mechanism in SDP negotiator, mainly for specific formats that require SDP fmtp negotiation.
- Modified video codec ID string to use encoding name+payload type (was encoding name+clock rate), also added encoding description in video codec info, so duplicated codecs (e.g: multiple H264 configurations) can be differentiated.
- Few enhancements for H264 in ffmpeg wrapper (e.g: added proper profile-id & packetization-mode setup).
git-svn-id: https://svn.pjsip.org/repos/pjproject/branches/projects/2.0-dev@3500 74dad513-b988-da41-8d7b-12977e46ad98
- Initial version of video stream integration into pjsua-lib.
- Replaced audio info array in pjsua_call_info with media info array.
- Added video media info into call dump.
- Fixed assertion caused by pjsua_set_state(NULL) logging after pjlib shutdown.
git-svn-id: https://svn.pjsip.org/repos/pjproject/branches/projects/2.0-dev@3463 74dad513-b988-da41-8d7b-12977e46ad98
1185: Dynamic creation of media transports
============================================
Done:
- media transports are created on demand now
Todo:
- media transport creation is still blocking
1201: Video support in PJSUA-LIB
===================================
Done:
- call now supports N media (N audio and M video)
- number of audio/video streams is configurable per acc
- extra audio stream info in pjsua_call_info to support multiple audio streams
in one call
- video subsys and ffmpeg initialization in PJSUA-LIB
- ability to offer and create video SDP answer
- "dq" for more than 1 audio streams
- introducing pjsua_state and pjsua_get_state()
API change:
- on_stream_created() and on_stream_destroyed() callbacks: changed session to
stream
Todo:
- many others features are disabled, just search for DISABLED_FOR_TICKET_1185
macro (these have also been added to ticket #1193 (Issues & Todos)). Notable
missing features are:
- creation of duplicate SDP m= lines for optional SRTP
- mm.. that's it?
- whole lot of testings
pjsua:
===============
- Added --extra-audio and --video options. Specify these more than once and
each time an extra audio/video streams will be added. :)
git-svn-id: https://svn.pjsip.org/repos/pjproject/branches/projects/2.0-dev@3457 74dad513-b988-da41-8d7b-12977e46ad98
- Updated pj_register_strerror() to just return PJ_SUCCESS when the same range
and handler is being re-registered.
- Removed the usage of static flag of error string handler registration in some
modules, which prevent the re-registration of the handler, e.g: in restarting
pjsua, as such flags never got reseted.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3455 74dad513-b988-da41-8d7b-12977e46ad98
* uri_test:
Fixes a divide by zero error when the benchmark is run on a really fast machine.
* presence:
Fixes a compiler warning about potential referencing of an uninitialized variable.
* echo_speex:
Allow for compilation when PJMEDIA_HAS_SPEEX_AEC is not defined.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3443 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3383 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3376 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3374 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3349 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3334 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3330 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3323 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3322 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3313 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3301 74dad513-b988-da41-8d7b-12977e46ad98
- incoming multipart message will be handled automatically
- for testing, enable HAVE_MULTIPART_TEST in pjsua_app.c
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3243 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3239 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3219 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3217 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3216 74dad513-b988-da41-8d7b-12977e46ad98
- 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: https://svn.pjsip.org/repos/pjproject/trunk@3215 74dad513-b988-da41-8d7b-12977e46ad98
- added new PJ_ICE_STRANS_OP_KEEP_ALIVE operation
- also added new on_ice_transport_error() pjsua callback to allow application to react to the failure.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3212 74dad513-b988-da41-8d7b-12977e46ad98
- Fixed invite module to reset SDP negotiator state after incomplete SDP offer-answer in re-INVITE/UPDATE.
- Added some SIPp test scenarios.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3208 74dad513-b988-da41-8d7b-12977e46ad98
- Added lock codec feature to make sure that only one codec is active, by updating media session using UPDATE (if remote supports it) or re-INVITE.
- Added few SIPp test scenarios.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3206 74dad513-b988-da41-8d7b-12977e46ad98
- Fixed compile error (on Symbian) incompatible types between int and pjsip_dialog_cap_status.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3200 74dad513-b988-da41-8d7b-12977e46ad98
- Fixed process_answer() of SDP negotiation, when no common format in a media, instead of returning error, it should just deactivate the media (offer & answer) and continue negotiating next media.
- Generalized the way of deactivating media: set port to 0 and remove all attributes.
- Added new API pjmedia_sdp_media_clone_deactivate() to clone media and deactivate the newly cloned media.
- Updated PJMEDIA SDP negotiation test.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3198 74dad513-b988-da41-8d7b-12977e46ad98
- Added a feature in dialog to store and retrieve remote capabilities dug from the remote messages.
- Added few APIs in dialog to query and update remote capabilities, also added an API in pjsua_call to query whether a capability is supported by remote.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3196 74dad513-b988-da41-8d7b-12977e46ad98
- Moved auto reregistration scheduling to be before the registration callback.
- Updated validations in auto_rereg_timer_cb() & schedule_reregistration().
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3185 74dad513-b988-da41-8d7b-12977e46ad98
- Added new approach of SRTP optional mode in pjsua-lib by duplicating SDP media line for secured and unsecured version of media transport.
- Integrated this feature into pjsua app, it is activated via --use-srtp=3 param.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3172 74dad513-b988-da41-8d7b-12977e46ad98
- removed the assertion (allow responding to empty realm)
- slight modification in Authenticate/WWW-Authenticate headers to allow printing challenge with empty realm, otherwise a malformed header will be printed if empty realm is given
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3149 74dad513-b988-da41-8d7b-12977e46ad98
- link error undefined reference to `.L23' in function pjsip_cred_info_cmp() (thanks Ken Fish for the report).
- compile error unable to convert pj_uint32_t to pjsip_transport_state.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3148 74dad513-b988-da41-8d7b-12977e46ad98
- Moved the code of disconnecting calls (after first re-reg attempt failure) to schedule_reregistration(), so it will be executed earlier (right after the failure of re-reg attempt).
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3139 74dad513-b988-da41-8d7b-12977e46ad98
- Added functions to set/unset transport state notification callback on specific transport.
- Updated transaction to immediately terminate the transactions when their transport gets disconnected.
- Minor update: renamed function pjsip_tpmgr_set/get_status_cb() to pjsip_tpmgr_set/get_state_cb().
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3138 74dad513-b988-da41-8d7b-12977e46ad98
- Added initial version of automatic re-registration after registration failure and automatic call disconnection after re-registration attempt fails.
- Published auto re-registration setting to pjsua app.
- Updated pjsip_regc_send() to retrieve the transport earlier (was only in tsx_callback()).
- Fixed TCP and TLS transport to prevent transport deletion in transport disconnection callback.
- Fixed wrong keep-alive settings used by TLS transport (was using TCP keep-alive settings).
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3128 74dad513-b988-da41-8d7b-12977e46ad98
- PJSUA-LIB transport callback, if installed, will call the previously registered callback, to allow multiple transport callbacks to be installed
- there seem to be a bug with the use of "pjsip_tp_state_callback" everywhere (the "pjsip_tp_state_callback" type is pointer, but most variables of this type are declared to pointer too)
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3119 74dad513-b988-da41-8d7b-12977e46ad98
- Updated transport state notification callback to return void.
- Updated transport state enum to only contain connected and disconnected, no more bitmask value.
- Added direction field to SIP transport.
- Removed remote hostname hash from transport key.
- Updated cert info dump to return -1 when buffer is insufficient.
- Added new error code PJSIP_TLS_ECERTVERIF.
- Updated get_cert_name() in ssl_sock_symbian.c to use heap buffer instead of stack.
- Minors, e.g: added prefix PJ in cipher types, docs.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3110 74dad513-b988-da41-8d7b-12977e46ad98
- Initial version of server domain name verification:
- Updated SSL certificate info, especially identities info
- Updated verification mechanism as in the specifications in ticket desc.
- Added server domain name info in pjsip_tx_data.
- Added alternative API for acquiring transport and creating transport of transport factory to include pjsip_tx_data param.
- Server identity match criteria:
- full host name match
- wild card not accepted
- if identity is URI, it must be SIP/SIPS URI
- Initial version of transport state notifications:
- Added new API to set transport state callback in PJSIP and PJSUA.
- Defined states: connected/disconnected, accepted/rejected, verification errors.
- Minors:
- Updated SSL socket test: dump verification result, test of requiring client cert, and few minors.
- Updated test cert to include subjectAltName extensions.
- Added SSL certificate dump function.
- Updated max number of socket async operations in Symbian sample apps (RSocketServ::Connect()) to 32 (was default 8).
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3106 74dad513-b988-da41-8d7b-12977e46ad98
- implemented in sip_reg.c instead of in PJSUA-LIB, so that the functionality can be reused by non-PJSUA-LIB applications
- also added several Python test scripts
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3105 74dad513-b988-da41-8d7b-12977e46ad98
- in this implementation, when pjsua_acc_set_registration(FALSE) is called, the un-REGISTER request will be sent immediately after un-PUBLISH, unlike the process during shutdown where the un-REGISTER request will be sent only after un-PUBLISH transaction is complete
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3096 74dad513-b988-da41-8d7b-12977e46ad98
- fixed the problem that caused ACK not to be sent. This happened when TCP switching is used, and the TCP fails to send the request.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3090 74dad513-b988-da41-8d7b-12977e46ad98
- fix for r3071: added protection for case when TSX_HAS_PENDING_TRANSPORT flag is set to the transaction but pending_tx is NULL, causing crash
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3084 74dad513-b988-da41-8d7b-12977e46ad98
- Added new API pjmedia_codec_mgr_set_default_param() to set/update default codec parameter and implemented pjsua_codec_set_param() based on it.
- Added mutex in codec manager to protect states manipulations.
- Modified API pjmedia_codec_mgr_init() to add pool factory param.
- Added new API pjmedia_codec_mgr_destroy().
- Updated passthrough codec AMR to regard peer's mode-set setting.
- Fixed VAS audio device to apply AMR encoding bitrate setting.
- Fixed IPP codec codec_open() to update AMR bitrate info (for stream) when AMR encoding bitrate is not using the default, e.g: requested by peer via format param 'mode-set' in SDP.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3074 74dad513-b988-da41-8d7b-12977e46ad98
- adhere to --prefix
- header and lib files installation
- pkgconfig creation
- also added version.mak to fill in with the correct version
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@3063 74dad513-b988-da41-8d7b-12977e46ad98
Nanang Izzuddin
Benny Prijono
Sauw Ming
Perry Ismangil