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res_rtp_asterisk: Don't send RTP before DTLS has negotiated.

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There was no check in __rtp_sendto that prevented Asterisk from sending
RTP before DTLS had finished negotiating. This patch adds logic to do
so.

Fixes: #1260
This commit is contained in:
Ben Ford
2025-08-04 12:22:12 -05:00
committed by github-actions[bot]
parent 57344a8061
commit adde97bff3
1 changed files with 10 additions and 0 deletions
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10
res/res_rtp_asterisk.c
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@@ -3451,6 +3451,16 @@ static int __rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t siz
struct ast_rtp *transport_rtp = ast_rtp_instance_get_data(transport);
struct ast_srtp *srtp = ast_rtp_instance_get_srtp(transport, rtcp);
int res;
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
char *out = buf;
struct dtls_details *dtls = !rtcp ? &rtp->dtls : &rtp->rtcp->dtls;
/* Don't send RTP if DTLS hasn't finished yet */
if (dtls->ssl && ((*out < 20) || (*out > 63)) && dtls->connection == AST_RTP_DTLS_CONNECTION_NEW) {
*via_ice = 0;
return 0;
}
#endif
*via_ice = 0;