diff options
| author | Mark Michelson <mmichelson@digium.com> | 2016-05-26 15:14:50 -0500 |
|---|---|---|
| committer | Richard Mudgett <rmudgett@digium.com> | 2016-05-27 11:00:09 -0500 |
| commit | bb0f4a6310fecb0f160b9e481fa751de6e35724a (patch) | |
| tree | 9008fe54580c7a4d0026f72233dd08a4385192c9 /channels/chan_rtp.c | |
| parent | a6b16d7029a7a44c6a43b3b0f1cbc74cecd11454 (diff) | |
multicast RTP: Add dialing options
This adds a new parameter to the end of a multicast RTP dialing string.
This parameter defines the following options:
* i: Set the interface from which multicast RTP is sent
* l: Set whether multicast packets are looped back to the sender
* t: Set the TTL for multicast packets
* c: Set the codec to use for RTP
ASTERISK-26068 #close
Reported by Mark Michelson
Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219
Diffstat (limited to 'channels/chan_rtp.c')
| -rw-r--r-- | channels/chan_rtp.c | 17 |
1 files changed, 15 insertions, 2 deletions
diff --git a/channels/chan_rtp.c b/channels/chan_rtp.c index 56705b1f0..093602823 100644 --- a/channels/chan_rtp.c +++ b/channels/chan_rtp.c @@ -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"); @@ -163,7 +166,15 @@ 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", args.destination); @@ -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", @@ -207,12 +218,14 @@ 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; } |