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git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@4387 74dad513-b988-da41-8d7b-12977e46ad98
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incoming call and fixed typo in assertion in sip_inv.c
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@4067 74dad513-b988-da41-8d7b-12977e46ad98
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callbacks when updating 'pjsua_var' is needed, while updating 'pjsua_call' should be enough with call/dialog lock (which is actually being held by the INVITE session layer during invoking its callback).
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3977 74dad513-b988-da41-8d7b-12977e46ad98
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challenged with authentication (thanks Bogdan Krakowski for the fix)
git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3954 74dad513-b988-da41-8d7b-12977e46ad98
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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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git-svn-id: http://svn.pjsip.org/repos/pjproject/branches/1.x@3768 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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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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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/trunk@3553 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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- 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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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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- 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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- 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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- 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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- 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: http://svn.pjsip.org/repos/pjproject/trunk@3206 74dad513-b988-da41-8d7b-12977e46ad98
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- 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: http://svn.pjsip.org/repos/pjproject/trunk@3196 74dad513-b988-da41-8d7b-12977e46ad98
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instead if it is greater
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3013 74dad513-b988-da41-8d7b-12977e46ad98
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before media channel initialization.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2955 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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- fixed misc compiler warnings with gcc on Linux
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2938 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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- Renamed pjsip_timer_default_setting() to pjsip_timer_setting_default().
- Updated session timer settings in pjsua-lib as whole session timer setting struct (pyhton version remains using se & min_se).
- Added output param SIP status code in pjsip_timer_process_resp() and pjsip_timer_process_req() to specify the corresponding SIP status code when function returning non-PJ_SUCCESS.
- Fixed print header functions in sip_timer.c to have buffer check.
- Added PJSIP_SESS_TIMER_DEF_SE setting to specify the default value of session timer interval.
- Fixed role reference of the refresher, it is transaction role, not dialog role.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2859 74dad513-b988-da41-8d7b-12977e46ad98
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- Initial version of Session Timers (RFC 4028).
- Added new options in pjsua app to configure Session Timers settings.
- Added python tests for Session Timers.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2858 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2752 74dad513-b988-da41-8d7b-12977e46ad98
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- will send SUBSCRIBE to refresh REFER subscription (not REFER!), only when required (such as when call transfer is running for longer than REFER subscription expiration, hence need to be refreshed)
- replaced hardcoded REFER subscription duration (600s) with a macro, {{{PJSIP_XFER_EXPIRES}}}.
- when NOTIFY with "200 OK" sipfrag body is received and subscription state is not terminated, send SUBSCRIBE with Expires=0 to terminate the REFER subscription
- for transferee, terminate the subscription in CONNECTING state and not in CONFIRMED state. Terminating the subscription in CONFIRMED state causes redundant NOTIFYs with "200 OK" sipfrag body to be sent, one with active subscription and another with terminated state.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2750 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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transfer scenario (thanks Tomáš Valenta for the report!)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2695 74dad513-b988-da41-8d7b-12977e46ad98
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Joel Dodson for the report)
- the INVITE session now correctly uses the SDP offer "fixed" by the negotiator, hence it will have the correct origin fields.
- removed update_sdp_version() from PJSUA-LIB
- the negotiator now also fixes the session ID of subsequent answer so that it's identical to the previous SDP
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2643 74dad513-b988-da41-8d7b-12977e46ad98
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(thanks Rostislav Molodyko for the report)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2593 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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well
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2394 74dad513-b988-da41-8d7b-12977e46ad98
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fixed various transmit data buffer leaks when transmission fails immediately
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2380 74dad513-b988-da41-8d7b-12977e46ad98
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it more natural to use
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2371 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2370 74dad513-b988-da41-8d7b-12977e46ad98
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INVITE are received due to forking
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2315 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2301 74dad513-b988-da41-8d7b-12977e46ad98
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- Changed rem_rtp/rtcp_addr to src_rtp/rtcp_addr in media transport info
- Updated behaviour of media transport get info, when the transport hasn't receive any packet src_rtp/rtcp_addr will not be set.
- Fixed bug in pjsua_call_dump that rem_addr was unset.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2293 74dad513-b988-da41-8d7b-12977e46ad98
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- Added new fields in media transport info: remote address originating RTP & RTCP.
- Updated transport UDP & ICE to fill above fields in getting transport info.
- Updated pjsua call dump, instead of showing remote RTP address from SDP, it will show address of RTP originator.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@2291 74dad513-b988-da41-8d7b-12977e46ad98
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