chan_rtp.c: Simplify options to UnicastRTP channel creation.

Change the awkward and not as flexible UnicastRTP options format
From:
Dial(UnicastRTP/127.0.0.1[/[<engine>][/[<codec>]]])
To:
Dial(UnicastRTP/127.0.0.1[/[<options>]])

Where <options> can be standard Asterisk flag options:
c(<codec>) - Specify which codec/format to use such as 'ulaw'.
e(<engine>) - Specify which RTP engine to use such as 'asterisk'.

More option flags can be easily added later such as the codec's RTP
payload type to use when the codec does not have a static payload type
defined.

Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9

2 files changed, 69 insertions, 10 deletions

diff --git a/CHANGES b/CHANGES
index 608a4a4b3..e799f71ef 100644
--- a/CHANGES
+++ b/CHANGES
@@ -135,6 +135,32 @@ chan_iax2
    seconds. Setting this to a higher value may help in lagged networks or those
    experiencing high packet loss.
 
+chan_rtp (was chan_multicast_rtp)
+------------------
+ * Added unicast RTP support and renamed chan_multicast_rtp to chan_rtp.
+
+ * The format for dialing a unicast RTP channel is:
+   UnicastRTP/<destination-addr>[/[<options>]]
+   Where <destination-addr> is something like '127.0.0.1:5060'.
+   Where <options> are in standard Asterisk flag options format:
+   c(<codec>) - Specify which codec/format to use such as 'ulaw'.
+   e(<engine>) - Specify which RTP engine to use such as 'asterisk'.
+
+ * New options were added for a multicast RTP channel.  The format for
+   dialing a multicast RTP channel is:
+   MulticastRTP/<type>/<destination-addr>[/[<control-addr>][/[<options>]]]
+   Where <type> can be either 'basic' or 'linksys'.
+   Where <destination-addr> is something like '224.0.0.3:5060'.
+   Where <control-addr> is something like '127.0.0.1:5060'.
+   Where <options> are in standard Asterisk flag options format:
+   c(<codec>) - Specify which codec/format to use such as 'ulaw'.
+   i(<address>) - Specify the interface address from which multicast RTP
+     is sent.
+   l(<enable>) - Set whether packets are looped back to the sender.  The
+     enable value can be 0 to set looping to off and non-zero to set
+     looping on.
+   t(<ttl>) - Set the time-to-live (TTL) value for multicast packets.
+
 chan_sip
 ------------------
  * New 'rtpbindaddr' global setting. This allows a user to define which
diff --git a/channels/chan_rtp.c b/channels/chan_rtp.c
index 093602823..0fe66bd20 100644
--- a/channels/chan_rtp.c
+++ b/channels/chan_rtp.c
@@ -176,7 +176,7 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
 		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;
 	}
@@ -230,6 +230,25 @@ 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)
 {
@@ -240,11 +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");
@@ -262,17 +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;
 		}
@@ -283,12 +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;
 	}