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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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@@ -12775,7 +12775,13 @@ static void add_ice_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a
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while ((candidate = ao2_iterator_next(&i))) {
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ast_str_append(a_buf, 0, "a=candidate:%s %d %s %d ", candidate->foundation, candidate->id, candidate->transport, candidate->priority);
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ast_str_append(a_buf, 0, "%s ", ast_sockaddr_stringify_host(&candidate->address));
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ast_str_append(a_buf, 0, "%s typ ", ast_sockaddr_stringify_port(&candidate->address));
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if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_SRFLX
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&& candidate->id == AST_RTP_ICE_COMPONENT_RTCP) {
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ast_str_append(a_buf, 0, "%d typ ", ast_sockaddr_port(&candidate->address) + 1);
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} else {
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ast_str_append(a_buf, 0, "%s typ ", ast_sockaddr_stringify_port(&candidate->address));
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}
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if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_HOST) {
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ast_str_append(a_buf, 0, "host");
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