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chan_rtp.c: Simplify options to UnicastRTP channel creation.

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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
changes/56/2956/1
Richard Mudgett 9 years ago
parent 3e8d523d88
commit dca052e531
2 changed files with 69 additions and 10 deletions
Show Stats Download Patch File Download Diff File

26
CHANGES
Unescape View File

@ -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

53
channels/chan_rtp.c
Unescape View File

@ -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;
}

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