diff --git a/CHANGES b/CHANGES
index 608a4a4b33b3176baddd114bee3227dc3f4afc7e..e799f71ef8b378fb22e4d0aa9eaa0141304f9a36 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 093602823d4d865ec4ff45114b9ceb32f3541773..0fe66bd2092a0e121770c606c35a98bf5e5345ef 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;
}