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Merge "res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup"

This commit is contained in:
George Joseph 2020-02-17 11:28:07 -06:00 committed by Gerrit Code Review
parent 12ba0682ed 3865b3fd6a
commit 1fc1336b2c
1 changed files with 28 additions and 2 deletions
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30
res/res_rtp_asterisk.c
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@ -2480,6 +2480,9 @@ static void dtls_perform_handshake(struct ast_rtp_instance *instance, struct dtl
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
ast_debug(3, "dtls_perform_handshake (%p) - ssl = %p, setup = %d\n",
rtp, dtls->ssl, dtls->dtls_setup);
/* If we are not acting as a client connecting to the remote side then
* don't start the handshake as it will accomplish nothing and would conflict
* with the handshake we receive from the remote side.
@ -2516,6 +2519,8 @@ static void dtls_perform_setup(struct dtls_details *dtls)
SSL_set_connect_state(dtls->ssl);
}
dtls->connection = AST_RTP_DTLS_CONNECTION_NEW;
ast_debug(3, "dtls_perform_setup - connection reset'\n");
}
#endif
@ -2548,11 +2553,23 @@ static void ast_rtp_on_ice_complete(pj_ice_sess *ice, pj_status_t status)
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
dtls_perform_setup(&rtp->dtls);
ast_debug(3, "ast_rtp_on_ice_complete (%p) - perform DTLS\n", rtp);
/*
* Seemingly no reason to call dtls_perform_setup here. Currently we'll do a full
* protocol level renegotiation if things do change. And if bundled is being used
* then ICE is reused when a stream is added.
*
* Note, if for some reason in the future dtls_perform_setup does need to done here
* be aware that creates a race condition between the call here (on ice completion)
* and potential DTLS handshaking when receiving RTP. What happens is the ssl object
* can get cleared (SSL_clear) during that handshaking process (DTLS init). If that
* happens then Asterisk won't complete DTLS initialization. RTP packets are still
* sent/received but won't be encrypted/decrypted.
*/
dtls_perform_handshake(instance, &rtp->dtls, 0);
if (rtp->rtcp && rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_STANDARD) {
dtls_perform_setup(&rtp->rtcp->dtls);
dtls_perform_handshake(instance, &rtp->rtcp->dtls, 1);
}
#endif
@ -2868,6 +2885,8 @@ static int dtls_srtp_setup(struct ast_rtp *rtp, struct ast_rtp_instance *instanc
struct dtls_details *dtls = !rtcp ? &rtp->dtls : &rtp->rtcp->dtls;
int index;
ast_debug(3, "Setup DTLS SRTP (%p)'\n", rtp);
/* If a fingerprint is present in the SDP make sure that the peer certificate matches it */
if (rtp->dtls_verify & AST_RTP_DTLS_VERIFY_FINGERPRINT) {
X509 *certificate;
@ -2906,6 +2925,7 @@ static int dtls_srtp_setup(struct ast_rtp *rtp, struct ast_rtp_instance *instanc
}
if (dtls_srtp_add_local_ssrc(rtp, instance, rtcp, ast_rtp_instance_get_ssrc(instance), 1)) {
ast_log(LOG_ERROR, "Failed to add local source '%p'\n", rtp);
return -1;
}
@ -3014,6 +3034,8 @@ static int __rtp_recvfrom(struct ast_rtp_instance *instance, void *buf, size_t s
return -1;
}
ast_debug(3, "__rtp_recvfrom (%p) - Got SSL packet '%d'\n", rtp, *in);
/*
* A race condition is prevented between dtls_perform_handshake()
* and this function because both functions have to get the
@ -3053,6 +3075,8 @@ static int __rtp_recvfrom(struct ast_rtp_instance *instance, void *buf, size_t s
}
/* Notify that dtls has been established */
res = RTP_DTLS_ESTABLISHED;
ast_debug(3, "__rtp_recvfrom (%p) - DTLS established'\n", rtp);
} else {
/* Since we've sent additional traffic start the timeout timer for retransmission */
dtls_srtp_start_timeout_timer(instance, rtp, rtcp);
@ -8519,6 +8543,8 @@ static int ast_rtp_activate(struct ast_rtp_instance *instance)
}
#endif
ast_debug(3, "ast_rtp_activate (%p) - setup and perform DTLS'\n", rtp);
dtls_perform_setup(&rtp->dtls);
dtls_perform_handshake(instance, &rtp->dtls, 0);