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version 1.x
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@4428 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@4426 74dad513-b988-da41-8d7b-12977e46ad98
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changed but ICE transport is running.
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@4346 74dad513-b988-da41-8d7b-12977e46ad98
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media is unchanged.
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@4338 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@4337 74dad513-b988-da41-8d7b-12977e46ad98
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set to PJ_TRUE and fixed the string duplication of encoding name
Thanks to Hideo and Fredrik for the reports.
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@4334 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@4329 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@4261 74dad513-b988-da41-8d7b-12977e46ad98
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argument are ignored
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@4065 74dad513-b988-da41-8d7b-12977e46ad98
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- static reference counter for PJLIB init/shutdown.
- implemented atexit() in PJMEDIA and PJSIP level: pjmedia_endpt_atexit() & pjsip_endpt_atexit().
- updated pjmedia/transport_srtp.c, pjsip/sip_timer.c, and pjsip/sip_replaces.c to use the new atexit() functions.
- API change: pjmedia_srtp_init_lib() now requires 'pjmedia_endpt' param.
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3986 74dad513-b988-da41-8d7b-12977e46ad98
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find_audio_index() in pjsua_media_channel_create_sdp(), so find_audio_index() will also verify the media count in the remote SDP.
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3962 74dad513-b988-da41-8d7b-12977e46ad98
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device
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3913 74dad513-b988-da41-8d7b-12977e46ad98
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without sending any outgoing messages
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3829 74dad513-b988-da41-8d7b-12977e46ad98
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Arrhenius for the contribution!)
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3816 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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stream.
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3571 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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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3438 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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- 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 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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- 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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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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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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allocation once if TURN allocation fails. If this allocation retry also fails, notify the TURN user via on_ice_complete() callback. Details:
- 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: http://svn.pjsip.org/repos/pjproject/trunk@3212 74dad513-b988-da41-8d7b-12977e46ad98
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- 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: http://svn.pjsip.org/repos/pjproject/trunk@3198 74dad513-b988-da41-8d7b-12977e46ad98
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- 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: http://svn.pjsip.org/repos/pjproject/trunk@3172 74dad513-b988-da41-8d7b-12977e46ad98
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PJSUA-LIB destroy sequence)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3153 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3078 74dad513-b988-da41-8d7b-12977e46ad98
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build errors (thanks Michael Bradley Jr for the report)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3075 74dad513-b988-da41-8d7b-12977e46ad98
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- 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: http://svn.pjsip.org/repos/pjproject/trunk@3074 74dad513-b988-da41-8d7b-12977e46ad98
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sockets have been resolved, so reduce chattiness during initialization and simplify debugging related to STUN problems
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3041 74dad513-b988-da41-8d7b-12977e46ad98
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hold-resume (thanks Nikolay Popok for the report).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2996 74dad513-b988-da41-8d7b-12977e46ad98
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- implementation:
- PJLIB (sock_qos*.*)
- added QoS support in:
- SIP UDP transport,
- SIP TCP transport,
- media UDP transport (done in pjsua-lib),
- pjnath ICE stream transport,
- pjnath STUN socket,
- pjnath TURN client
- added QoS options in pjsua-lib:
- QoS fields in pjsua_transport_config
- added "--set-qos" parameter in pjsua
Notes:
- QoS in TLS transport is not yet implemented, waiting for #957
- build ok on VS6, VS2005 (multiple targets), Carbide, and Mingw
- no run-time testing yet
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2966 74dad513-b988-da41-8d7b-12977e46ad98
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candidate has changed
- done
- added pj_ice_strans_state (to be used for informational purposes for now)
- added pjmedia ICE transport specific info, and display it in call dump output
- misc fixes (changed pjmedia_transport_info.spec_info_cnt from int to unsigned)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2945 74dad513-b988-da41-8d7b-12977e46ad98
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- wait for unregistration to complete (or a preconfigured delay expires)
- new account config field to set the maximum delay to wait for unregistration
- rejects incoming requests (INVITE, SUBSCRIBE, and OPTIONS) when shutdown is in progress
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2943 74dad513-b988-da41-8d7b-12977e46ad98
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call is running for long period of time and with many re-INVITES
- introducing flip-flop pools in the pjsip_inv_session. There are two additional pools created, and one of them will be reset everytime SDP negotiation is done to release memory back to the OS
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2869 74dad513-b988-da41-8d7b-12977e46ad98
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more robustness, and continue application startup if STUN resolution fails
PJSUA-LIB:
- New fields in pjsua_config to specify more than one STUN servers (the stun_srv_cnt and stun_srv array)
- The existing stun_host and stun_domain fields are deprecated, but backward compatibility is maintained. If stun_srv_cnt is zero, the library will import the entries from stun_host and stun_domain
- The library will now resolve the STUN server entries one by one and test it before using it
- New auxiliary API pjsua_resolve_stun_servers() to perform resolution and test against array of STUN servers
pjsua application:
- The "stun-srv" command line options can now be specified more than once
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2864 74dad513-b988-da41-8d7b-12977e46ad98
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called, even after pjsua_set_snd_dev() is called
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2837 74dad513-b988-da41-8d7b-12977e46ad98
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- Added default ilbc mode into codec passthrough setting.
- Added iLBC mode 'negotiation' in iLBC codec_open().
- Updated stream_create() to prioritize codec_open(), that may update the codec params, over stream initializations involving codec params.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2834 74dad513-b988-da41-8d7b-12977e46ad98
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- Added a new API pjmedia_codec_passthrough_init2().
- Updated the initialization steps of passthrough codec in pjsua_media.c, to configure the codecs (of passthrough codec) to be enabled based on audio device extended/encoded formats.
- Minor update: added passthrough.h into pjmedia_codec.vcproj.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2825 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2758 74dad513-b988-da41-8d7b-12977e46ad98
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- fixed crash when null-audio is used with switchboard
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2742 74dad513-b988-da41-8d7b-12977e46ad98
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--null-audio is set
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2741 74dad513-b988-da41-8d7b-12977e46ad98
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- #793: AMR encoder should regard 'mode-set' param specified by remote decoder.
- #831: Automatically switch to TCP transport when sending large request
- #832: Support for outbound proxy setting without using Route header
- #849: Modify conference audio switch behavior in connecting ports.
- #850: Remove 'Require=replaces' param in 'Refer-To' header (in call transfer with replaces).
- #851: Support for regular nomination in ICE
- #852: --ip-addr support for IPv6 for media transport in pjsua
- #854: Adding SOFTWARE attribute in all outgoing requests may cause compatibility problem with older STUN server (thanks Alexei Kuznetsov for the report)
- #855: Bug in digit map frequencies for DTMF digits (thanks FCCH for the report)
- #856: Put back the ICE candidate priority values according to the default values in the draft-mmusic-ice
- #857: Support for ICE keep-alive with Binding indication
- #858: Do not authenticate STUN 438 response
- #859: AMR-WB format param in the SDP is not negotiated correctly.
- #867: Return error instead of asserting when PJSUA-LIB fails to open log file
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2724 74dad513-b988-da41-8d7b-12977e46ad98
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device to not be opened
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2679 74dad513-b988-da41-8d7b-12977e46ad98
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initialization.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2597 74dad513-b988-da41-8d7b-12977e46ad98
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- Initial source of G.722.1/Annex C integration.
- Disabled some "odd" modes of L16 codec (11kHz & 22kHz mono & stereo) while releasing some payload types.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2563 74dad513-b988-da41-8d7b-12977e46ad98
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aps-direct branch to trunk.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2506 74dad513-b988-da41-8d7b-12977e46ad98
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which is called each time a call being disconnected (for any reason).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2425 74dad513-b988-da41-8d7b-12977e46ad98
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