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chan_sip: Fix RTCP port for SRFLX ICE candidates

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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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@402345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kinsey Moore
2013-11-01 12:31:49 +00:00
parent 45a6969b3c
commit 2d5debf45c
3 changed files with 28 additions and 19 deletions
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8
channels/chan_sip.c
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@@ -12616,7 +12616,13 @@ static void add_ice_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a
while ((candidate = ao2_iterator_next(&i))) {
ast_str_append(a_buf, 0, "a=candidate:%s %d %s %d ", candidate->foundation, candidate->id, candidate->transport, candidate->priority);
ast_str_append(a_buf, 0, "%s ", ast_sockaddr_stringify_host(&candidate->address));
ast_str_append(a_buf, 0, "%s typ ", ast_sockaddr_stringify_port(&candidate->address));
if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_SRFLX
&& candidate->id == AST_RTP_ICE_COMPONENT_RTCP) {
ast_str_append(a_buf, 0, "%d typ ", ast_sockaddr_port(&candidate->address) + 1);
} else {
ast_str_append(a_buf, 0, "%s typ ", ast_sockaddr_stringify_port(&candidate->address));
}
if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_HOST) {
ast_str_append(a_buf, 0, "host");