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authorJoshua Colp <jcolp@digium.com>2012-09-20 18:27:28 +0000
committerJoshua Colp <jcolp@digium.com>2012-09-20 18:27:28 +0000
commite8380afc8a147ee299c3881423b2e0b27c4cfc0d (patch)
tree9930ca060cafb0821bd7f2d977f1aede33a67877 /channels/sip/include
parentf1fb120f5d62933cac50dc47810418ebf535af7c (diff)
Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.
As mentioned on the review for this, WebRTC has moved towards choosing DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds support for this but makes it available for normal SIP clients as well. Testing has been done to ensure that this introduces no regressions with existing behavior and also that it functions as expected. Review: https://reviewboard.asterisk.org/r/2113/ ........ Merged revisions 373229 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels/sip/include')
-rw-r--r--channels/sip/include/sip.h5
1 files changed, 5 insertions, 0 deletions
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 047310fc4..48a5a3bb5 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -36,6 +36,7 @@
#include "asterisk/security_events.h"
#include "asterisk/features.h"
#include "asterisk/http_websocket.h"
+#include "asterisk/rtp_engine.h"
#ifndef FALSE
#define FALSE 0
@@ -1212,6 +1213,8 @@ struct sip_pvt {
struct ast_cc_config_params *cc_params;
struct sip_epa_entry *epa_entry;
int fromdomainport; /*!< Domain port to show in from field */
+
+ struct ast_rtp_dtls_cfg dtls_cfg;
};
/*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission
@@ -1361,6 +1364,8 @@ struct sip_peer {
enum sip_peer_type type; /*!< Distinguish between "user" and "peer" types. This is used solely for CLI and manager commands */
unsigned int disallowed_methods;
struct ast_cc_config_params *cc_params;
+
+ struct ast_rtp_dtls_cfg dtls_cfg;
};
/*!