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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.
This commit is contained in:
Maximilian Fridrich 2023-09-05 09:32:53 +02:00 committed by asterisk-org-access-app[bot]
parent 4db98a38f1
commit 0f416925db
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");