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chan_sip: Fix RTCP port for SRFLX ICE candidates
This corrects one-way audio between Asterisk and Chrome/jssip as a result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX ICE candidates. This also exposes an ICE component enumeration to extract further details from candidates. (closes issue ASTERISK-21383) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/2967/ ........ Merged revisions 402345 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402348 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -381,10 +381,16 @@ enum ast_rtp_ice_candidate_type {
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AST_RTP_ICE_CANDIDATE_TYPE_RELAYED, /*!< ICE relayed candidate, which represents the address allocated in TURN server. */
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};
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/*! \brief ICE component types */
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enum ast_rtp_ice_component_type {
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AST_RTP_ICE_COMPONENT_RTP = 1,
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AST_RTP_ICE_COMPONENT_RTCP = 2,
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};
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/*! \brief Structure for an ICE candidate */
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struct ast_rtp_engine_ice_candidate {
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char *foundation; /*!< Foundation identifier */
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unsigned int id; /*!< Component identifier */
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enum ast_rtp_ice_component_type id; /*!< Component identifier */
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char *transport; /*!< Transport for the media */
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int priority; /*!< Priority which is used if multiple candidates can be used */
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struct ast_sockaddr address; /*!< Address of the candidate */
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