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pjsip: fix support for allow=all
This change adds improvements to support for allow=all in pjsip.conf so that it functions as intended. Previously, the allow/disallow socery configuration would set & clear codecs from the media.codecs and media.prefs list, but if all was specified the prefs list was not updated. Then a call would fail when create_outgoing_sdp_stream() created an SDP with no audio codecs. A new function ast_codec_pref_append_all() is provided to add all codecs to the prefs list - only those not already on the list. This enables the configuration to specify a codec preference, but still add all codecs, and even then remove some codecs, as shown in this example: allow = ulaw, alaw, all, !g729, !g723 Also, the display order of allow in cli output is updated to match the configuration by using prefs instead of caps when generating a human readable string. Finally, a change to create_outgoing_sdp_stream() skips a codec when it does not have a payload code instead of the call failing. (closes issue ASTERISK-23018) Reported by: xrobau Review: https://reviewboard.asterisk.org/r/3131/ ........ Merged revisions 405875 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405876 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- include/asterisk/format_pref.h 3 additions, 0 deletionsinclude/asterisk/format_pref.h
- main/format_pref.c 36 additions, 0 deletionsmain/format_pref.c
- main/frame.c 2 additions, 0 deletionsmain/frame.c
- main/sorcery.c 4 additions, 2 deletionsmain/sorcery.c
- res/res_pjsip_sdp_rtp.c 2 additions, 1 deletionres/res_pjsip_sdp_rtp.c
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