[asterisk-commits] chan sip: Don't refuse calls with "optional crypto"; fall ba... (asterisk[13])

[SVN commits to the Asterisk project]

Anonymous Coward #1000019 has submitted this change and it was merged.

Change subject: chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP.
......................................................................


chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP.

Certain SNOM phones send so-called "optional crypto" in their SDP body.
Regular SRTP setup looks like this:

    m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101
    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:...

SNOM-style "optional crypto" looks like this:

    m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101
    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:...

A crypto line is supplied, but the m-line does not have SAVP.

When res_srtp.so is *not* loaded, then chan_sip.so treats the optional
crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the
incoming call with the following message:

    WARNING: process_sdp: Failed to receive SDP offer/answer with
    required SRTP crypto attributes for audio

For platforms that want to start providing SRTP this presents a
compatibility problem.

This changeset lets chan_sip handle the SDP as if no crypto-line was
supplied: i.e. accept the call as regular RTP, just like it did before
res_srtp was loaded.

Now you'll get this informative warning instead:

    WARNING: Ignoring crypto attribute in SDP because RTP transport is
    insecure

ASTERISK-23989 #close
Reported by: Olle Johansson

Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2
---
M channels/chan_sip.c
1 file changed, 24 insertions(+), 5 deletions(-)

Approvals:
  George Joseph: Looks good to me, but someone else must approve
  Anonymous Coward #1000019: Verified
  Joshua Colp: Looks good to me, approved

Objections:
  Alexander Traud: I would prefer this is not merged as is



diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index c0aae46..c8987a0 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1477,7 +1477,8 @@
 static void handle_response(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
 
 /*------ SRTP Support -------- */
-static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp, const char *a);
+static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp,
+		const char *a, int secure_transport);
 
 /*------ T38 Support --------- */
 static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
@@ -10618,7 +10619,7 @@
 						}
 					} else if (process_sdp_a_sendonly(value, &sendonly)) {
 						processed = TRUE;
-					} else if (!processed_crypto && process_crypto(p, p->rtp, &p->srtp, value)) {
+					} else if (!processed_crypto && process_crypto(p, p->rtp, &p->srtp, value, secure_audio)) {
 						processed_crypto = TRUE;
 						processed = TRUE;
 					} else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) {
@@ -10635,7 +10636,7 @@
 						if (p->vsrtp) {
 							ast_set_flag(p->vsrtp, AST_SRTP_CRYPTO_OFFER_OK);
 						}
-					} else if (!processed_crypto && process_crypto(p, p->vrtp, &p->vsrtp, value)) {
+					} else if (!processed_crypto && process_crypto(p, p->vrtp, &p->vsrtp, value, secure_video)) {
 						processed_crypto = TRUE;
 						processed = TRUE;
 					} else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) {
@@ -10648,7 +10649,7 @@
 						processed = TRUE;
 					} else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
 						processed = TRUE;
-					} else if (!processed_crypto && process_crypto(p, p->trtp, &p->tsrtp, value)) {
+					} else if (!processed_crypto && process_crypto(p, p->trtp, &p->tsrtp, value, 1)) {
 						processed_crypto = TRUE;
 						processed = TRUE;
 					}
@@ -33640,7 +33641,8 @@
 	ao2_iterator_destroy(&iter);
 }
 
-static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp, const char *a)
+static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp,
+		const char *a, int secure_transport)
 {
 	struct ast_rtp_engine_dtls *dtls;
 
@@ -33656,6 +33658,23 @@
 	/* skip "crypto:" */
 	a += strlen("crypto:");
 
+	if (!secure_transport) {
+		/* > The Secure Real-time Transport Protocol (SRTP)
+		 * > [RFC3711] provides security services for RTP media
+		 * > and is signaled by use of secure RTP transport (e.g.,
+		 * > "RTP/SAVP" or "RTP/SAVPF") in an SDP media (m=) line.
+		 * > ...
+		 * > The "crypto" attribute MUST only appear at the SDP
+		 * > media level (not at the session level).
+		 *
+		 * Ergo, we can trust RTP/(S)AVP to be read from the m=
+		 * line before we get here. If it was RTP/AVP, then this
+		 * is SNOM-specific optional SRTP. Ignore it.
+		 */
+		ast_log(LOG_WARNING, "Ignoring crypto attribute in SDP because RTP transport is insecure\n");
+		return FALSE;
+	}
+
 	if (!*srtp) {
 		if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
 			ast_log(LOG_WARNING, "Ignoring unexpected crypto attribute in SDP answer\n");

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Gerrit-MessageType: merged
Gerrit-Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Walter Doekes <walter+asterisk at wjd.nu>
Gerrit-Reviewer: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>