diff options
| author | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-06-08 05:29:08 +0000 |
|---|---|---|
| committer | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-06-08 05:29:08 +0000 |
| commit | 9b1a36a294342fc418d9a359a4cf06bd90c4acb9 (patch) | |
| tree | ecc27fc0db142ea1cd335a74cd1265f993fecd11 /include/asterisk/rtp_engine.h | |
| parent | 5f87b66641d86dbe7afec3b083016b2b1aceafc7 (diff) | |
Add SRTP support for Asterisk
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: http://svn.digium.com/svn/asterisk/trunk@268894 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'include/asterisk/rtp_engine.h')
| -rw-r--r-- | include/asterisk/rtp_engine.h | 11 |
1 files changed, 11 insertions, 0 deletions
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h index c7b433972..7d55a85f7 100644 --- a/include/asterisk/rtp_engine.h +++ b/include/asterisk/rtp_engine.h @@ -71,6 +71,9 @@ extern "C" { #include "asterisk/astobj2.h" #include "asterisk/frame.h" +#include "asterisk/netsock.h" +#include "asterisk/sched.h" +#include "asterisk/res_srtp.h" /* Maximum number of payloads supported */ #define AST_RTP_MAX_PT 256 @@ -458,6 +461,11 @@ int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *m */ int ast_rtp_engine_unregister(struct ast_rtp_engine *engine); +int ast_rtp_engine_register_srtp(struct ast_srtp_res *srtp_res, struct ast_srtp_policy_res *policy_res); + +void ast_rtp_engine_unregister_srtp(void); +int ast_rtp_engine_srtp_is_registered(void); + #define ast_rtp_glue_register(glue) ast_rtp_glue_register2(glue, ast_module_info->self) /*! @@ -1724,6 +1732,9 @@ struct ast_rtp_glue *ast_rtp_instance_get_active_glue(struct ast_rtp_instance *i */ struct ast_channel *ast_rtp_instance_get_chan(struct ast_rtp_instance *instance); +int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *policy); +struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance); + #if defined(__cplusplus) || defined(c_plusplus) } #endif |