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Add SRTP support for Asterisk

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After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.

Original patch by mikma, updated for trunk and revised by me.

(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel

Review: https://reviewboard.asterisk.org/r/191/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Terry Wilson
2010-06-08 05:29:08 +00:00
parent ebbf166c2d
commit 857814f435
28 changed files with 9227 additions and 30793 deletions
Download Patch File Download Diff File Expand all files Collapse all files

9
channels/sip/include/sip.h
View File

@@ -307,10 +307,8 @@
#define SIP_PAGE2_RTAUTOCLEAR (1 << 1) /*!< GP: Should we clean memory from peers after expiry? */
#define SIP_PAGE2_RPID_UPDATE (1 << 2)
#define SIP_PAGE2_Q850_REASON (1 << 3) /*!< DP: Get/send cause code via Reason header */
#define SIP_PAGE2_SYMMETRICRTP (1 << 4) /*!< GDP: Whether symmetric RTP is enabled or not */
#define SIP_PAGE2_STATECHANGEQUEUE (1 << 5) /*!< D: Unsent state pending change exists */
#define SIP_PAGE2_CONNECTLINEUPDATE_PEND (1 << 6)
#define SIP_PAGE2_RPID_IMMEDIATE (1 << 7)
#define SIP_PAGE2_RPORT_PRESENT (1 << 8) /*!< Was rport received in the Via header? */
@@ -345,6 +343,7 @@
#define SIP_PAGE2_UDPTL_DESTINATION (1 << 25) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 26) /*!< DP: Always set up video, even if endpoints don't support it */
#define SIP_PAGE2_HAVEPEERCONTEXT (1 << 27) /*< Are we associated with a configured peer context? */
#define SIP_PAGE2_USE_SRTP (1 << 28) /*!< DP: Whether we should offer (only) SRTP */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
@@ -352,7 +351,7 @@
SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC | \
SIP_PAGE2_RPID_IMMEDIATE | SIP_PAGE2_RPID_UPDATE | SIP_PAGE2_SYMMETRICRTP |\
SIP_PAGE2_Q850_REASON | SIP_PAGE2_HAVEPEERCONTEXT)
SIP_PAGE2_Q850_REASON | SIP_PAGE2_HAVEPEERCONTEXT | SIP_PAGE2_USE_SRTP)
#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
@@ -965,6 +964,7 @@ struct sip_pvt {
* or respect the other endpoint's request for frame sizes (on)
* for incoming calls
*/
unsigned short req_secure_signaling:1;/*!< Whether we are required to have secure signaling or not */
char tag[11]; /*!< Our tag for this session */
int timer_t1; /*!< SIP timer T1, ms rtt */
int timer_b; /*!< SIP timer B, ms */
@@ -1048,6 +1048,9 @@ struct sip_pvt {
AST_LIST_HEAD_NOLOCK(request_queue, sip_request) request_queue; /*!< Requests that arrived but could not be processed immediately */
struct sip_invite_param *options; /*!< Options for INVITE */
struct sip_st_dlg *stimer; /*!< SIP Session-Timers */
struct sip_srtp *srtp; /*!< Structure to hold Secure RTP session data for audio */
struct sip_srtp *vsrtp; /*!< Structure to hold Secure RTP session data for video */
struct sip_srtp *tsrtp; /*!< Structure to hold Secure RTP session data for text */
int red; /*!< T.140 RTP Redundancy */
int hangupcause; /*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */