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.

diff --git a/channels/chan_rtp.c b/channels/chan_rtp.c
index 0740c2c6a16d12df327fee655c436f88ae7f1621..5d2c282ad3c7f97d79a58b86bce988879d8b887c 100644 (file)
--- a/channels/chan_rtp.c
+++ b/channels/chan_rtp.c
@@ -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");