Asterisk - The Open Source Telephony Project GIT-master-f36a736
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Data Fields | |
unsigned char | buf [8192+AST_FRIENDLY_OFFSET] |
const struct ast_srtp_cb * | cb |
void * | data |
struct ao2_container * | policies |
unsigned char | rtcpbuf [8192+AST_FRIENDLY_OFFSET] |
struct ast_rtp_instance * | rtp |
srtp_t | session |
int | warned |
Definition at line 66 of file res_srtp.c.
unsigned char buf[8192+AST_FRIENDLY_OFFSET] |
Definition at line 73 of file res_srtp.c.
Referenced by ast_srtp_protect().
const struct ast_srtp_cb* cb |
Definition at line 70 of file res_srtp.c.
Referenced by ast_srtp_set_cb(), and ast_srtp_unprotect().
void* data |
Definition at line 71 of file res_srtp.c.
Referenced by ast_srtp_set_cb(), ast_srtp_unprotect(), and srtp_event_cb().
struct ao2_container* policies |
Definition at line 68 of file res_srtp.c.
Referenced by ast_srtp_add_stream(), ast_srtp_destroy(), ast_srtp_unprotect(), find_policy(), and res_srtp_new().
unsigned char rtcpbuf[8192+AST_FRIENDLY_OFFSET] |
Definition at line 74 of file res_srtp.c.
Referenced by ast_srtp_protect().
struct ast_rtp_instance* rtp |
Definition at line 67 of file res_srtp.c.
Referenced by ast_rtp_change_source(), ast_srtp_create(), ast_srtp_replace(), ast_srtp_unprotect(), and calculate_lost_packet_statistics().
srtp_t session |
Definition at line 69 of file res_srtp.c.
Referenced by ast_srtp_add_stream(), ast_srtp_change_source(), ast_srtp_create(), ast_srtp_destroy(), ast_srtp_protect(), and ast_srtp_unprotect().
int warned |
Definition at line 72 of file res_srtp.c.
Referenced by ast_srtp_unprotect(), and res_srtp_new().