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Merge "chan_rtp.c: Simplify options to UnicastRTP channel creation."

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This commit is contained in:
Joshua Colp
2016-06-08 05:13:59 -05:00
committed by Gerrit Code Review
parent 71f70f5a07 dca052e531
commit 5164e1fd85
2 changed files with 69 additions and 10 deletions
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26
CHANGES
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@@ -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
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;
}