[asterisk-bugs] [JIRA] (ASTERISK-22961) [patch] DTLS-SRTP not working with SHA-256

[Juan Ramirez (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-22961?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=218566#comment-218566 ] 

Juan Ramirez edited comment on ASTERISK-22961 at 5/28/14 3:48 PM:
------------------------------------------------------------------

@Nitesh

There's a problem with the patch you've uploaded, take a look a the following text I've extracted from the file:
{noformat}
 	if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT)) {
@@ -5977,14 +5974,15 @@
 			ice->stop(dialog->trtp);
 		}

-		if (dialog_initialize_dtls_srtp(dialog, dialog->trtp, &dialog->tsrtp)) {
-			return -1;
-		}
-
 		/* Do not timeout text as its not constant*/
 		ast_rtp_instance_set_keepalive(dialog->trtp, dialog->rtpkeepalive);

 		ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
+
+		/*SRTP_DTLS */
+		if (dialog_initialize_dtls_srtp(dialog, dialog->vrtp, &dialog->vsrtp)) {
+			return -1;
+		}
 	}
{noformat}

It is initializing the *video* RTP instead of the *text* rtp

BTW, I'm merging both patches into my Asterisk 11.9 branch and I have seen that they have similiar modifications, I'll upload the results as soon as I can.

Regards

Juan



was (Author: ichramm):
@Nitesh

There's a problem with the patch you've uploaded, take a look a the following text extrated from the file:
{noformat}
 	if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT)) {
@@ -5977,14 +5974,15 @@
 			ice->stop(dialog->trtp);
 		}

-		if (dialog_initialize_dtls_srtp(dialog, dialog->trtp, &dialog->tsrtp)) {
-			return -1;
-		}
-
 		/* Do not timeout text as its not constant*/
 		ast_rtp_instance_set_keepalive(dialog->trtp, dialog->rtpkeepalive);

 		ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
+
+		/*SRTP_DTLS */
+		if (dialog_initialize_dtls_srtp(dialog, dialog->vrtp, &dialog->vsrtp)) {
+			return -1;
+		}
 	}
{noformat}

It is initializing the *video* RTP instead of the *text* rtp

BTW, I'm merging both patches into my Asterisk 11.9 branch and I have seen that they have similiar modifications, I'll upload the results as soon as I can.

Regards

Juan


> [patch] DTLS-SRTP not working with SHA-256
> ------------------------------------------
>
>                 Key: ASTERISK-22961
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22961
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Channels/chan_sip/SRTP, Channels/chan_sip/WebSocket
>    Affects Versions: 11.6.0, 12.0.0-beta2
>            Reporter: Jay Jideliov
>         Attachments: 11.7 patched.zip, asterisk_dtls.patch, backtrace (1).txt, chan_sip.c, dtls_retransmission.patch, ice_session.c, jssip no ring.txt, res_rtp_asterisk.c, res_rtp_asterisk.c, srtp_dtls.patch, srtp_dtls.patch, srtp_dtls.patch, wireshark.txt
>
>
> Recently it became possible to use websocket on asterisk without a proxy previously necessary to make calls from the web browser. Although partial support has been added, full browser cross-operability has not been achieved yet. However, it seems to be a relatively easy task.
> Tested on Chrome+SIPML5+Asterisk 11, the connection can be established and works fine. However, due to the fact that Firefox sends SHA-256 packets which are not supported by asterisk, hence the support for this browser is limited by this issue.
> Step 1: Adding certificates to support DTLS
> dtlsenable = yes
> dtlsverify = no
> dtlscertfile=/etc/asterisk/keys/softphone.pem
> dtlsprivatekey=/etc/asterisk/keys/key.pem
> dtlscafile=/etc/asterisk/keys/key.pem
> Step 2: Making a call
> [Nov 25 15:05:50] WARNING[5628][C-0000005c]: chan_sip.c:11034 process_sdp_a_dtls: Unsupported fingerprint hash type 'sha-2' received on dialog '38f43a1f-15cd-ad69-c2b3-72c21b9de5fd'



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