chan_pjsip: add a new function PJSIP_DTMF_MODE

This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis

ASTERISK-27085 #close

Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612

3 files changed, 192 insertions, 2 deletions

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index c542e149d..db0a69712 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -1701,7 +1701,7 @@ static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
 	struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
 	int res = 0;
 
-	switch (channel->session->endpoint->dtmf) {
+	switch (channel->session->dtmf) {
 	case AST_SIP_DTMF_RFC_4733:
 		if (!media || !media->rtp) {
 			return -1;
@@ -1821,7 +1821,7 @@ static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned in
 	struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
 	int res = 0;
 
-	switch (channel->session->endpoint->dtmf) {
+	switch (channel->session->dtmf) {
 	case AST_SIP_DTMF_AUTO_INFO:
 	{
 		if (!media || !media->rtp) {
@@ -2633,6 +2633,12 @@ static struct ast_custom_function media_offer_function = {
 	.write = pjsip_acf_media_offer_write
 };
 
+static struct ast_custom_function dtmf_mode_function = {
+	.name = "PJSIP_DTMF_MODE",
+	.read = pjsip_acf_dtmf_mode_read,
+	.write = pjsip_acf_dtmf_mode_write
+};
+
 static struct ast_custom_function session_refresh_function = {
 	.name = "PJSIP_SEND_SESSION_REFRESH",
 	.write = pjsip_acf_session_refresh_write,
@@ -2677,6 +2683,11 @@ static int load_module(void)
 		goto end;
 	}
 
+	if (ast_custom_function_register(&dtmf_mode_function)) {
+		ast_log(LOG_WARNING, "Unable to register PJSIP_DTMF_MODE dialplan function\n");
+		goto end;
+	}
+
 	if (ast_custom_function_register(&session_refresh_function)) {
 		ast_log(LOG_WARNING, "Unable to register PJSIP_SEND_SESSION_REFRESH dialplan function\n");
 		goto end;
@@ -2736,6 +2747,7 @@ static int load_module(void)
 end:
 	ao2_cleanup(pjsip_uids_onhold);
 	pjsip_uids_onhold = NULL;
+	ast_custom_function_unregister(&dtmf_mode_function);
 	ast_custom_function_unregister(&media_offer_function);
 	ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
 	ast_custom_function_unregister(&session_refresh_function);
@@ -2758,6 +2770,7 @@ static int unload_module(void)
 	ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
 	ast_sip_session_unregister_supplement(&call_pickup_supplement);
 
+	ast_custom_function_unregister(&dtmf_mode_function);
 	ast_custom_function_unregister(&media_offer_function);
 	ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
 	ast_custom_function_unregister(&session_refresh_function);
diff --git a/channels/pjsip/dialplan_functions.c b/channels/pjsip/dialplan_functions.c
index 22d773820..a382c4e0a 100644
--- a/channels/pjsip/dialplan_functions.c
+++ b/channels/pjsip/dialplan_functions.c
@@ -68,6 +68,18 @@
 		<ref type="function">PJSIP_SEND_SESSION_REFRESH</ref>
 	</see-also>
 </function>
+<function name="PJSIP_DTMF_MODE" language="en_US">
+	<synopsis>
+		Get or change the DTMF mode for a SIP call.
+	</synopsis>
+	<syntax>
+	</syntax>
+	<description>
+		<para>When read, returns the current DTMF mode</para>
+		<para>When written, sets the current DTMF mode</para>
+		<para>This function uses the same DTMF mode naming as the dtmf_mode configuration option</para>
+	</description>
+</function>
 <function name="PJSIP_SEND_SESSION_REFRESH" language="en_US">
 	<synopsis>
 		W/O: Initiate a session refresh via an UPDATE or re-INVITE on an established media session
@@ -440,6 +452,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/app.h"
 #include "asterisk/channel.h"
 #include "asterisk/format.h"
+#include "asterisk/dsp.h"
 #include "asterisk/pbx.h"
 #include "asterisk/res_pjsip.h"
 #include "asterisk/res_pjsip_session.h"
@@ -1039,6 +1052,34 @@ int pjsip_acf_media_offer_write(struct ast_channel *chan, const char *cmd, char
 	return ast_sip_push_task_synchronous(channel->session->serializer, media_offer_write_av, &mdata);
 }
 
+int pjsip_acf_dtmf_mode_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
+{
+	struct ast_sip_channel_pvt *channel;
+
+	if (!chan) {
+		ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
+		return -1;
+	}
+
+	ast_channel_lock(chan);
+	if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
+		ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
+		ast_channel_unlock(chan);
+		return -1;
+	}
+
+	channel = ast_channel_tech_pvt(chan);
+
+	if (ast_sip_dtmf_to_str(channel->session->dtmf, buf, len) < 0) {
+		ast_log(LOG_WARNING, "Unknown DTMF mode %d on PJSIP channel %s\n", channel->session->dtmf, ast_channel_name(chan));
+		ast_channel_unlock(chan);
+		return -1;
+	}
+
+	ast_channel_unlock(chan);
+	return 0;
+}
+
 struct refresh_data {
 	struct ast_sip_session *session;
 	enum ast_sip_session_refresh_method method;
@@ -1067,6 +1108,117 @@ static int sip_session_response_cb(struct ast_sip_session *session, pjsip_rx_dat
 	return 0;
 }
 
+static int dtmf_mode_refresh_cb(void *obj)
+{
+	struct refresh_data *data = obj;
+
+	if (data->session->inv_session->state == PJSIP_INV_STATE_CONFIRMED) {
+		ast_debug(3, "Changing DTMF mode on channel %s after OFFER/ANSER completion. Sending session refresh\n", ast_channel_name(data->session->channel));
+
+		ast_sip_session_refresh(data->session, NULL, NULL,
+			sip_session_response_cb, data->method, 1);
+	}
+
+	return 0;
+}
+
+int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
+{
+	struct ast_sip_channel_pvt *channel;
+	struct chan_pjsip_pvt *pjsip_pvt;
+	int dsp_features = 0;
+	int dtmf = -1;
+	struct refresh_data rdata = {
+			.method = AST_SIP_SESSION_REFRESH_METHOD_INVITE,
+		};
+
+	if (!chan) {
+		ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
+		return -1;
+	}
+
+	ast_channel_lock(chan);
+	if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
+		ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
+		ast_channel_unlock(chan);
+		return -1;
+	}
+
+	channel = ast_channel_tech_pvt(chan);
+	rdata.session = channel->session;
+
+	dtmf = ast_sip_str_to_dtmf(value);
+
+	if (dtmf == -1) {
+		ast_log(LOG_WARNING, "Cannot set DTMF mode to '%s' on channel '%s' as value is invalid.\n", value,
+			ast_channel_name(chan));
+		ast_channel_unlock(chan);
+		return -1;
+	}
+
+	if (channel->session->dtmf == dtmf) {
+		/* DTMF mode unchanged, nothing to do! */
+		ast_channel_unlock(chan);
+		return 0;
+	}
+
+	channel->session->dtmf = dtmf;
+
+	pjsip_pvt = channel->pvt;
+	if (pjsip_pvt->media[SIP_MEDIA_AUDIO]  && (pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp) {
+		if (channel->session->dtmf == AST_SIP_DTMF_RFC_4733) {
+			ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 1);
+			ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_RFC2833);
+		} else if (channel->session->dtmf == AST_SIP_DTMF_INFO) {
+			ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
+			ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_NONE);
+		} else if (channel->session->dtmf == AST_SIP_DTMF_INBAND) {
+			ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
+			ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_INBAND);
+		} else if (channel->session->dtmf == AST_SIP_DTMF_NONE) {
+			ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
+			ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_NONE);
+		} else if (channel->session->dtmf == AST_SIP_DTMF_AUTO) {
+			if (ast_rtp_instance_dtmf_mode_get((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp) != AST_RTP_DTMF_MODE_RFC2833) {
+				/* no RFC4733 negotiated, enable inband */
+				ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_INBAND);
+			}
+		} else if (channel->session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
+			ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
+			if (ast_rtp_instance_dtmf_mode_get((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp) == AST_RTP_DTMF_MODE_INBAND) {
+				/* if inband, switch to INFO */
+				ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_NONE);
+			}
+		}
+	}
+
+	if (channel->session->dsp) {
+		dsp_features = ast_dsp_get_features(channel->session->dsp);
+	}
+	if (channel->session->dtmf == AST_SIP_DTMF_INBAND ||
+		channel->session->dtmf == AST_SIP_DTMF_AUTO) {
+		dsp_features |= DSP_FEATURE_DIGIT_DETECT;
+	} else {
+		dsp_features &= ~DSP_FEATURE_DIGIT_DETECT;
+	}
+	if (dsp_features) {
+		if (!channel->session->dsp) {
+			if (!(channel->session->dsp = ast_dsp_new())) {
+				ast_channel_unlock(chan);
+				return 0;
+			}
+		}
+		ast_dsp_set_features(channel->session->dsp, dsp_features);
+	} else if (channel->session->dsp) {
+		ast_dsp_free(channel->session->dsp);
+		channel->session->dsp = NULL;
+	}
+
+	ast_channel_unlock(chan);
+
+	return ast_sip_push_task_synchronous(channel->session->serializer, dtmf_mode_refresh_cb, &rdata);
+}
+
 static int refresh_write_cb(void *obj)
 {
 	struct refresh_data *data = obj;
diff --git a/channels/pjsip/include/dialplan_functions.h b/channels/pjsip/include/dialplan_functions.h
index 8b80bfa74..731e91d13 100644
--- a/channels/pjsip/include/dialplan_functions.h
+++ b/channels/pjsip/include/dialplan_functions.h
@@ -48,6 +48,31 @@ int pjsip_acf_channel_read(struct ast_channel *chan, const char *cmd, char *data
 int pjsip_acf_media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);
 
 /*!
+ * \brief PJSIP_DTMF_MODE function read callback
+ * \param chan The channel the function is called on
+ * \param cmd The name of the function
+ * \param data Arguments passed to the function
+ * \param buf Out buffer that should be populated with the data
+ * \param len Size of the buffer
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_acf_dtmf_mode_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
+
+/*!
+ * \brief PJSIP_DTMF_MODE function write callback
+ * \param chan The channel the function is called on
+ * \param cmd The name of the function
+ * \param data Arguments passed to the function
+ * \param value Value to be set by the function
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);
+
+/*!
  * \brief PJSIP_MEDIA_OFFER function read callback
  * \param chan The channel the function is called on
  * \param cmd The name of the function