Age | Commit message (Collapse) | Author |
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bypass_srtp, in transport_encode_sdp(), as it shouldn't change the media session states.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3384 74dad513-b988-da41-8d7b-12977e46ad98
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- updated the releasing of the old pool to be done after the new codec param is copied
- fixed the double dec_fmtp copy loop, one of them should be enc_fmtp copy loop instead
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3380 74dad513-b988-da41-8d7b-12977e46ad98
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pointer for the stream info source.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3379 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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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3362 74dad513-b988-da41-8d7b-12977e46ad98
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offer (thanks Marcus Froeschl for the suggestion))
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3360 74dad513-b988-da41-8d7b-12977e46ad98
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MAX_BURST_MSEC should be converted to number-of-frame unit (thanks Zhefeng Du for the fix).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3359 74dad513-b988-da41-8d7b-12977e46ad98
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- using mutex in accessing application callback pointers
- releasing mutex before calling application callbacks to avoid deadlock
- refine the synchronization of backend/libsrtp states
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3348 74dad513-b988-da41-8d7b-12977e46ad98
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- Added check in processing answer, if media offer port is zero, just skip negotiation process.
- Added SIPp test scenario.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3347 74dad513-b988-da41-8d7b-12977e46ad98
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coreaudio_dev
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3335 74dad513-b988-da41-8d7b-12977e46ad98
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according to padding length.
Payload padding in outgoing RTP investigation results:
- the RTP does not specify RTP payload alignment.
- most codecs also do not specify RTP payload alignment, usually only octet-alignment is specified and this seems to be done.
- SRTP, RFC3711 states:
- None of the pre-defined SRTP encryption transforms uses any padding; for these, the RTP and SRTP payload sizes match exactly.
- Message authentication codes define their own padding.
- Encryption transforms that use padding are vulnerable to subtle attacks, especially when message authentication is not used.
So, currently payload padding in outgoing RTP is not necessary.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3325 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3315 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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- Fixed bytes_per_frame calculation in stream port.
- Fixed sample streamutil.c to use codec info/param for codec bandwidth calculation (was using bytes_per_frame info of stream port).
- Doc fix for bytes_per_frame field in pjmedia_port_info.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3292 74dad513-b988-da41-8d7b-12977e46ad98
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status, disabled/off, after receiving 'fmtp:18 annexb=no' in the SDP.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3288 74dad513-b988-da41-8d7b-12977e46ad98
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pjmedia_codec_ipp_set/get_config() as currently to set PCM signal level adjustment can be done using the existing G722.1 API pjmedia_codec_g7221_set_pcm_shift().
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3263 74dad513-b988-da41-8d7b-12977e46ad98
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2005
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3262 74dad513-b988-da41-8d7b-12977e46ad98
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- Added PCM signal adjustment in IPP G722.1 implementation. The default setting is configurable via (the existing compile-time config) PJMEDIA_G7221_DEFAULT_PCM_SHIFT.
- Added new APIs to get and set IPP codecs settings: pjmedia_codec_ipp_set/get_config(). At run-time, the G722.1 PCM signal adjustment setting can be set using these functions.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3261 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3256 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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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3250 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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- Updated RTCP jitter statistics calculation (in receiving direction) to use "interarrival jitter" (was using "difference D") of RFC 3550.
- Added APIs to reset RTCP statistics.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3237 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3226 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed zeroed/unset RTP timestamp in RTCP sender report.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3224 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed SRTP transport to only (re)start the SRTP state when the SRTP crypto settings are updated.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3221 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3220 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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Johan Lantz for the suggestion)
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3218 74dad513-b988-da41-8d7b-12977e46ad98
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- 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: http://svn.pjsip.org/repos/pjproject/trunk@3217 74dad513-b988-da41-8d7b-12977e46ad98
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port clock when it is created with PJMEDIA_CLOCK_NO_ASYNC flag.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3214 74dad513-b988-da41-8d7b-12977e46ad98
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uninitialized memory ptr read under Valgrind. Thanks Jones Desougi for the patch.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3204 74dad513-b988-da41-8d7b-12977e46ad98
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- Added API pjmedia_codec_g722_set_pcm_shift() to enable configurable level-adjusment setting.
- Also added macro PJMEDIA_G722_DEFAULT_PCM_SHIFT (default value is 2) to accomplish 14-16 bit PCM conversion for G722 input/output.
- Added a feature in G722 to stop level-adjusment/PCM-shifting when clipping occured, compile-time configurable via PJMEDIA_G722_STOP_PCM_SHIFT_ON_CLIPPING macro.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3202 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed memory leak of CVoIPFormatIntfc instances in S60 VAS.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3201 74dad513-b988-da41-8d7b-12977e46ad98
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- Added new codec G721, as alias for G726-32.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3199 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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- Fixed SDP negotiation in processing answer: when the answer has less media count, generate disabled-media to match the media count.
- Added python test.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3195 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed SDP negotiation to ignore disabled media (with port 0) in the answer.
- Added a SIPp scenario for reproducing the issue.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3192 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3191 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed removing any unmatching formats in the remote-answer to also work with dynamic payload type.
- Updated reordering formats priority in the offer based on the answer to also work with dynamic payload type.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3187 74dad513-b988-da41-8d7b-12977e46ad98
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(from AMR payload) only when it is different from current encoding mode.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3184 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3175 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3174 74dad513-b988-da41-8d7b-12977e46ad98
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- Fixed max frames-per-packet constants in AMR codec (IPP and passthrough) to be based on PJMEDIA_MAX_FRAME_DURATION_MS setting.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3165 74dad513-b988-da41-8d7b-12977e46ad98
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hardcoded 1s, now it is 3/4 of JB max size and must not be lower than 1s.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3162 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3161 74dad513-b988-da41-8d7b-12977e46ad98
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- Added check if the negotiation result of local-offer/remote-answer has no matching format.
- Added routine to remove any unmatching formats in the remote-answer.
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3160 74dad513-b988-da41-8d7b-12977e46ad98
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git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3159 74dad513-b988-da41-8d7b-12977e46ad98
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- Removed orphaned third_party/gsm/inc/gsm.h.orig file
- Added support for external GSM header in /usr/include/gsm.h (rather than <gsm/gsm.h>)
Thanks Christopher Zimmermann for the fixes
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3158 74dad513-b988-da41-8d7b-12977e46ad98
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by possibility of uninitialized burst level after JB switches status INITIALIZING -> PROCESSING (thanks Janos Tolgyesi and Tamàs Solymosi for the report and investigation).
git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@3154 74dad513-b988-da41-8d7b-12977e46ad98
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