chan_sip: Do not send all codecs on INVITE.

Since version 13, Asterisk sent all allowed codecs as callee, even when the
caller did not request/support them. In case of dynamic RTP payloads, this led
to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the
intersection between the requested and the supported codecs is send again.

ASTERISK-24543 #close

Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 8f76e9c..f0d4de5 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -13332,7 +13332,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
                }
 
                /* Finally our remaining audio/video codecs */
-               for (x = 0; x < ast_format_cap_count(p->caps); x++) {
+               for (x = 0; ast_test_flag(&p->flags[0], SIP_OUTGOING) && x < ast_format_cap_count(p->caps); x++) {
                        tmp_fmt = ast_format_cap_get_format(p->caps, x);
 
                        if (ast_format_cap_iscompatible_format(alreadysent, tmp_fmt) != AST_FORMAT_CMP_NOT_EQUAL) {