[asterisk-bugs] [JIRA] (ASTERISK-29519) ROC value not incremented in SRTP

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29519?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=255608#comment-255608 ] 

Asterisk Team commented on ASTERISK-29519:
------------------------------------------

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> ROC value not incremented in SRTP
> ---------------------------------
>
>                 Key: ASTERISK-29519
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29519
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_srtp
>    Affects Versions: 18.4.0
>            Reporter: Marcos Cereijo Rodríguez
>
> h3. What is happening?
> We have to do an integration with a client that wants to secure their communications with TLS for the signaling and SRTP for the media.
> After enabling SRTP all looked great, but the client detected that after ~22 minutes the call hangup automatically.
> After some debugging, the client inform us that his SBC provider detected the issue. The reason of this problem was that after the RTP sequence number overflowed, the ROC (roll-over counter) value wasn't increased.
> h3. Proposed solution
> We had to update the `libsrtp` version from 2.0.0 to 2.3.0. The reason for this change is that the default version included in Debian doesn't expose the headers from reading and changing the values of ROC.
> For implementing this, we modified the following files:
> * res/res_rtp_asterisk.c
> * res/res_srtp.c
> The modifications for *res_srtp.c*:
> Lines 103-104
> {code}
> static int ast_srtp_set_stream_roc(struct ast_srtp *srtp, unsigned int ssrc, unsigned int roc);
> static int ast_srtp_get_stream_roc(struct ast_srtp *srtp, unsigned int ssrc, unsigned int *roc);
> {code}
> Lines 118-119
> {code}
> .set_stream_roc = ast_srtp_set_stream_roc,
> .get_stream_roc = ast_srtp_get_stream_roc
> {code}
> Lines 366-374
> {code}
> static int ast_srtp_set_stream_roc(struct ast_srtp *srtp, unsigned int ssrc, unsigned int roc)
> {
> 	return srtp_set_stream_roc(srtp->session, ssrc, roc);
> }
> static int ast_srtp_get_stream_roc(struct ast_srtp *srtp, unsigned int ssrc, unsigned int *roc)
> {
> 	return srtp_get_stream_roc(srtp->session, ssrc, roc);
> }
> {code}
> The modifications for *res_rtp_asterisk.c*:
> Lines 5099-5124
> {code}
> 	if (seqno == rtp->seqno) {
> 		rtp->seqno++;
> 		if (rtp->seqno == 0) {
> 			ast_verbose("Seq number roll over for SSRC: %d\n", rtp->ssrc);
> 			struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance, 0);
> 			if (res_srtp && srtp) {
> 				ast_verbose("\'SRTP\' detected\n");
> 				unsigned int roc;
> 				ast_verbose("Obtaining the SRTP ROC value...\n");
> 				int res = res_srtp->get_stream_roc(srtp, rtp->ssrc, &roc);
> 				if (res == 0) {
> 					ast_verbose("Current SRTP ROC value: %d", roc);
> 					roc++;
> 					ast_verbose("Setting new ROC to %d", roc);
> 					res = res_srtp->set_stream_roc(srtp, rtp->ssrc, roc);
> 					if (res == 0) {
> 						ast_verbose("ROC updated!\n");
> 					} else {
> 						ast_log(LOG_ERROR, "Cannot update ROC. Error: %d\n", res);
> 					}
> 				} else {
> 					ast_log(LOG_ERROR, "Cannot read current ROC. Error: %d\n", res);
> 				}
> 			}
> 		}
> 	}
> {code}



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