Merge "res_pjsip: Add fax_detect_timeout endpoint option."
[asterisk/asterisk.git] / channels / chan_rtp.c
index 56705b1..0fe66bd 100644 (file)
@@ -43,6 +43,7 @@ ASTERISK_REGISTER_FILE()
 #include "asterisk/rtp_engine.h"
 #include "asterisk/causes.h"
 #include "asterisk/format_cache.h"
+#include "asterisk/multicast_rtp.h"
 
 /* Forward declarations */
 static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
@@ -132,7 +133,9 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
                AST_APP_ARG(type);
                AST_APP_ARG(destination);
                AST_APP_ARG(control);
+               AST_APP_ARG(options);
        );
+       struct ast_multicast_rtp_options *mcast_options = NULL;
 
        if (ast_strlen_zero(data)) {
                ast_log(LOG_ERROR, "A multicast type and destination must be given to the 'MulticastRTP' channel\n");
@@ -163,9 +166,17 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
                goto failure;
        }
 
-       fmt = ast_format_cap_get_format(cap, 0);
+       mcast_options = ast_multicast_rtp_create_options(args.type, args.options);
+       if (!mcast_options) {
+               goto failure;
+       }
+
+       fmt = ast_multicast_rtp_options_get_format(mcast_options);
+       if (!fmt) {
+               fmt = ast_format_cap_get_format(cap, 0);
+       }
        if (!fmt) {
-               ast_log(LOG_ERROR, "No format available for sending RTP to '%s'\n",
+               ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
                        args.destination);
                goto failure;
        }
@@ -175,7 +186,7 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
                goto failure;
        }
 
-       instance = ast_rtp_instance_new("multicast", NULL, &control_address, args.type);
+       instance = ast_rtp_instance_new("multicast", NULL, &control_address, mcast_options);
        if (!instance) {
                ast_log(LOG_ERROR,
                        "Could not create '%s' multicast RTP instance for sending media to '%s'\n",
@@ -207,16 +218,37 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
 
        ao2_ref(fmt, -1);
        ao2_ref(caps, -1);
+       ast_multicast_rtp_free_options(mcast_options);
 
        return chan;
 
 failure:
        ao2_cleanup(fmt);
        ao2_cleanup(caps);
+       ast_multicast_rtp_free_options(mcast_options);
        *cause = AST_CAUSE_FAILURE;
        return NULL;
 }
 
+enum {
+       OPT_RTP_CODEC =  (1 << 0),
+       OPT_RTP_ENGINE = (1 << 1),
+};
+
+enum {
+       OPT_ARG_RTP_CODEC,
+       OPT_ARG_RTP_ENGINE,
+       /* note: this entry _MUST_ be the last one in the enum */
+       OPT_ARG_ARRAY_SIZE
+};
+
+AST_APP_OPTIONS(unicast_rtp_options, BEGIN_OPTIONS
+       /*! Set the codec to be used for unicast RTP */
+       AST_APP_OPTION_ARG('c', OPT_RTP_CODEC, OPT_ARG_RTP_CODEC),
+       /*! Set the RTP engine to use for unicast RTP */
+       AST_APP_OPTION_ARG('e', OPT_RTP_ENGINE, OPT_ARG_RTP_ENGINE),
+END_OPTIONS );
+
 /*! \brief Function called when we should prepare to call the unicast destination */
 static struct ast_channel *unicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
 {
@@ -227,11 +259,13 @@ static struct ast_channel *unicast_rtp_request(const char *type, struct ast_form
        struct ast_channel *chan;
        struct ast_format_cap *caps = NULL;
        struct ast_format *fmt = NULL;
+       const char *engine_name;
        AST_DECLARE_APP_ARGS(args,
                AST_APP_ARG(destination);
-               AST_APP_ARG(engine);
-               AST_APP_ARG(format);
+               AST_APP_ARG(options);
        );
+       struct ast_flags opts = { 0, };
+       char *opt_args[OPT_ARG_ARRAY_SIZE];
 
        if (ast_strlen_zero(data)) {
                ast_log(LOG_ERROR, "Destination is required for the 'UnicastRTP' channel\n");
@@ -249,17 +283,26 @@ static struct ast_channel *unicast_rtp_request(const char *type, struct ast_form
                goto failure;
        }
 
-       if (!ast_strlen_zero(args.format)) {
-               fmt = ast_format_cache_get(args.format);
+       if (!ast_strlen_zero(args.options)
+               && ast_app_parse_options(unicast_rtp_options, &opts, opt_args,
+                       ast_strdupa(args.options))) {
+               ast_log(LOG_ERROR, "'UnicastRTP' channel options '%s' parse error\n",
+                       args.options);
+               goto failure;
+       }
+
+       if (ast_test_flag(&opts, OPT_RTP_CODEC)
+               && !ast_strlen_zero(opt_args[OPT_ARG_RTP_CODEC])) {
+               fmt = ast_format_cache_get(opt_args[OPT_ARG_RTP_CODEC]);
                if (!fmt) {
-                       ast_log(LOG_ERROR, "Format '%s' not found for sending RTP to '%s'\n",
-                               args.format, args.destination);
+                       ast_log(LOG_ERROR, "Codec '%s' not found for sending RTP to '%s'\n",
+                               opt_args[OPT_ARG_RTP_CODEC], args.destination);
                        goto failure;
                }
        } else {
                fmt = ast_format_cap_get_format(cap, 0);
                if (!fmt) {
-                       ast_log(LOG_ERROR, "No format available for sending RTP to '%s'\n",
+                       ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
                                args.destination);
                        goto failure;
                }
@@ -270,12 +313,15 @@ static struct ast_channel *unicast_rtp_request(const char *type, struct ast_form
                goto failure;
        }
 
+       engine_name = S_COR(ast_test_flag(&opts, OPT_RTP_ENGINE),
+               opt_args[OPT_ARG_RTP_ENGINE], NULL);
+
        ast_ouraddrfor(&address, &local_address);
-       instance = ast_rtp_instance_new(args.engine, NULL, &local_address, NULL);
+       instance = ast_rtp_instance_new(engine_name, NULL, &local_address, NULL);
        if (!instance) {
                ast_log(LOG_ERROR,
                        "Could not create %s RTP instance for sending media to '%s'\n",
-                       S_OR(args.engine, "default"), args.destination);
+                       S_OR(engine_name, "default"), args.destination);
                goto failure;
        }