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chan_rtp: Implement RTP glue for UnicastRTP channels 

Resolves: #298

UserNote: The dial string option 'g' was added to the UnicastRTP channel
which enables RTP glue and therefore native RTP bridges with those
channels.

(cherry picked from commit 98ffcfebda)
This commit is contained in:
Maximilian Fridrich 2023-09-05 09:32:53 +02:00 committed by Asterisk Development Team
parent bc72c76891
commit 38598701da
1 changed files with 74 additions and 0 deletions
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74
channels/chan_rtp.c
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@ -249,6 +249,7 @@ failure:
enum {
OPT_RTP_CODEC = (1 << 0),
OPT_RTP_ENGINE = (1 << 1),
OPT_RTP_GLUE = (1 << 2),
};
enum {
@ -263,8 +264,14 @@ AST_APP_OPTIONS(unicast_rtp_options, BEGIN_OPTIONS
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),
/*! Provide RTP glue for the channel */
AST_APP_OPTION('g', OPT_RTP_GLUE),
END_OPTIONS );
static const struct ast_datastore_info chan_rtp_datastore_info = {
.type = "CHAN_RTP_GLUE",
};
/*! \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)
{
@ -372,6 +379,13 @@ static struct ast_channel *unicast_rtp_request(const char *type, struct ast_form
ast_channel_tech_set(chan, &unicast_rtp_tech);
if (ast_test_flag(&opts, OPT_RTP_GLUE)) {
struct ast_datastore *datastore;
if ((datastore = ast_datastore_alloc(&chan_rtp_datastore_info, NULL))) {
ast_channel_datastore_add(chan, datastore);
}
}
ast_format_cap_append(caps, fmt, 0);
ast_channel_nativeformats_set(chan, caps);
ast_channel_set_writeformat(chan, fmt);
@ -401,6 +415,61 @@ failure:
return NULL;
}
/*! \brief Function called by RTP engine to get peer capabilities */
static void chan_rtp_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
{
SCOPE_ENTER(1, "%s Native formats %s\n", ast_channel_name(chan),
ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(ast_channel_nativeformats(chan), &STR_TMP)));
ast_format_cap_append_from_cap(result, ast_channel_nativeformats(chan), AST_MEDIA_TYPE_UNKNOWN);
SCOPE_EXIT_RTN();
}
/*! \brief Function called by RTP engine to change where the remote party should send media.
*
* chan_rtp is not able to actually update the peer, so this function has no effect.
* */
static int chan_rtp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *tpeer, const struct ast_format_cap *cap, int nat_active)
{
return -1;
}
/*! \brief Function called by RTP engine to get local audio RTP peer */
static enum ast_rtp_glue_result chan_rtp_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
return AST_RTP_GLUE_RESULT_FORBID;
}
/*! \brief Function called by RTP engine to get local audio RTP peer */
static enum ast_rtp_glue_result chan_rtp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct ast_rtp_instance *rtp_instance = ast_channel_tech_pvt(chan);
struct ast_datastore *datastore;
if (!rtp_instance) {
return AST_RTP_GLUE_RESULT_FORBID;
}
if ((datastore = ast_channel_datastore_find(chan, &chan_rtp_datastore_info, NULL))) {
ao2_ref(datastore, -1);
*instance = rtp_instance;
ao2_ref(*instance, +1);
return AST_RTP_GLUE_RESULT_LOCAL;
}
return AST_RTP_GLUE_RESULT_FORBID;
}
/*! \brief Local glue for interacting with the RTP engine core */
static struct ast_rtp_glue unicast_rtp_glue = {
.type = "UnicastRTP",
.get_rtp_info = chan_rtp_get_rtp_peer,
.get_vrtp_info = chan_rtp_get_vrtp_peer,
.get_codec = chan_rtp_get_codec,
.update_peer = chan_rtp_set_rtp_peer,
};
/*! \brief Function called when our module is unloaded */
static int unload_module(void)
{
@ -412,6 +481,8 @@ static int unload_module(void)
ao2_cleanup(unicast_rtp_tech.capabilities);
unicast_rtp_tech.capabilities = NULL;
ast_rtp_glue_unregister(&unicast_rtp_glue);
return 0;
}
@ -421,6 +492,9 @@ static int load_module(void)
if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
return AST_MODULE_LOAD_DECLINE;
}
ast_rtp_glue_register(&unicast_rtp_glue);
ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
if (ast_channel_register(&multicast_rtp_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");