Support for negotiation and receiption of Cisco's RTP DTMF

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43597 f38db490-d61c-443f-a65b-d21fe96a405b

4 files changed, 111 insertions, 35 deletions

diff --git a/channels/h323/ast_h323.cxx b/channels/h323/ast_h323.cxx
index 61634aea9..be22ca64a 100644
--- a/channels/h323/ast_h323.cxx
+++ b/channels/h323/ast_h323.cxx
@@ -533,7 +533,7 @@ MyH323Connection::MyH323Connection(MyH323EndPoint & ep, unsigned callReference,
 	bridging = FALSE;
 	progressSetup = progressAlert = 0;
 	dtmfMode = 0;
-	dtmfCodec = (RTP_DataFrame::PayloadTypes)0;
+	dtmfCodec[0] = dtmfCodec[1] = (RTP_DataFrame::PayloadTypes)0;
 	redirect_reason = -1;
 #ifdef TUNNELLING
 	tunnelOptions = remoteTunnelOptions = 0;
@@ -664,7 +664,8 @@ void MyH323Connection::SetCallOptions(void *o, BOOL isIncoming)
 
 	progressSetup = opts->progress_setup;
 	progressAlert = opts->progress_alert;
-	dtmfCodec = (RTP_DataFrame::PayloadTypes)opts->dtmfcodec;
+	dtmfCodec[0] = (RTP_DataFrame::PayloadTypes)opts->dtmfcodec[0];
+	dtmfCodec[1] = (RTP_DataFrame::PayloadTypes)opts->dtmfcodec[1];
 	dtmfMode = opts->dtmfmode;
 
 	if (isIncoming) {
@@ -1213,7 +1214,8 @@ void MyH323Connection::SendUserInputTone(char tone, unsigned duration, unsigned
 
 void MyH323Connection::OnUserInputTone(char tone, unsigned duration, unsigned logicalChannel, unsigned rtpTimestamp)
 {
-	if ((dtmfMode & H323_DTMF_RFC2833)) {
+	/* Why we should check this? */
+	if ((dtmfMode & (H323_DTMF_CISCO | H323_DTMF_RFC2833 | H323_DTMF_SIGNAL)) != 0) {
 		if (h323debug) {
 			cout << "\t-- Received user input tone (" << tone << ") from remote" << endl;
 		}
@@ -1243,10 +1245,10 @@ void MyH323Connection::OnSendCapabilitySet(H245_TerminalCapabilitySet & pdu)
 			H245_Capability & cap = entry.m_capability;
 			if (cap.GetTag() == H245_Capability::e_receiveRTPAudioTelephonyEventCapability) {
 				H245_AudioTelephonyEventCapability & atec = cap;
-				atec.m_dynamicRTPPayloadType = dtmfCodec;
-//				on_set_rfc2833_payload(GetCallReference(), (const char *)GetCallToken(), (int)dtmfCodec);
+				atec.m_dynamicRTPPayloadType = dtmfCodec[0];
+//				on_set_rfc2833_payload(GetCallReference(), (const char *)GetCallToken(), (int)dtmfCodec[0]);
 				if (h323debug) {
-					cout << "\t-- Transmitting RFC2833 on payload " <<
+					cout << "\t-- Receiving RFC2833 on payload " <<
 						atec.m_dynamicRTPPayloadType << endl;
 				}
 			}
@@ -1299,21 +1301,12 @@ BOOL MyH323Connection::OnReceivedCapabilitySet(const H323Capabilities & remoteCa
 	};
 #endif
 	struct ast_codec_pref prefs;
+	RTP_DataFrame::PayloadTypes pt;
 
 	if (!H323Connection::OnReceivedCapabilitySet(remoteCaps, muxCap, reject)) {
 		return FALSE;
 	}
 
-	const H323Capability * cap = remoteCaps.FindCapability(H323_UserInputCapability::SubTypeNames[H323_UserInputCapability::SignalToneRFC2833]);
-	if (cap != NULL) {
-		RTP_DataFrame::PayloadTypes pt = ((H323_UserInputCapability*)cap)->GetPayloadType();
-		on_set_rfc2833_payload(GetCallReference(), (const char *)GetCallToken(), (int)pt);
-		if ((dtmfMode & H323_DTMF_RFC2833) && (sendUserInputMode == SendUserInputAsTone))
-			sendUserInputMode = SendUserInputAsInlineRFC2833;
-		if (h323debug) {
-			cout << "\t-- Inbound RFC2833 on payload " << pt << endl;
-		}
-	}
 	memset(&prefs, 0, sizeof(prefs));
 	int peer_capabilities = 0;
 	for (int i = 0; i < remoteCapabilities.GetSize(); ++i) {
@@ -1346,6 +1339,32 @@ BOOL MyH323Connection::OnReceivedCapabilitySet(const H323Capabilities & remoteCa
 				}
 			}
 			break;
+		case H323Capability::e_Data:
+			if (!strcmp((const char *)remoteCapabilities[i].GetFormatName(), CISCO_DTMF_RELAY)) {
+				pt = remoteCapabilities[i].GetPayloadType();
+				if ((dtmfMode & H323_DTMF_CISCO) != 0) {
+					on_set_rfc2833_payload(GetCallReference(), (const char *)GetCallToken(), (int)pt, 1);
+//					if (sendUserInputMode == SendUserInputAsTone)
+//						sendUserInputMode = SendUserInputAsInlineRFC2833;
+				}
+				if (h323debug) {
+					cout << "\t-- Outbound Cisco RTP DTMF on payload " << pt << endl;
+				}
+			}
+			break;
+		case H323Capability::e_UserInput:
+			if (!strcmp((const char *)remoteCapabilities[i].GetFormatName(), H323_UserInputCapability::SubTypeNames[H323_UserInputCapability::SignalToneRFC2833])) {
+				pt = remoteCapabilities[i].GetPayloadType();
+				if ((dtmfMode & H323_DTMF_RFC2833) != 0) {
+					on_set_rfc2833_payload(GetCallReference(), (const char *)GetCallToken(), (int)pt, 0);
+//					if (sendUserInputMode == SendUserInputAsTone)
+//						sendUserInputMode = SendUserInputAsInlineRFC2833;
+				}
+				if (h323debug) {
+					cout << "\t-- Outbound RFC2833 on payload " << pt << endl;
+				}
+			}
+			break;
 #if 0
 		case H323Capability::e_Video:
 			for (int x = 0; vcodecs[x].asterisk_codec > 0; ++x) {
@@ -1416,7 +1435,7 @@ BOOL MyH323Connection::OnStartLogicalChannel(H323Channel & channel)
 	return connectionState != ShuttingDownConnection;
 }
 
-void MyH323Connection::SetCapabilities(int cap, int dtmf_mode, void *_prefs, int pref_codec)
+void MyH323Connection::SetCapabilities(int caps, int dtmf_mode, void *_prefs, int pref_codec)
 {
 	PINDEX lastcap = -1; /* last common capability index */
 	int alreadysent = 0;
@@ -1427,11 +1446,12 @@ void MyH323Connection::SetCapabilities(int cap, int dtmf_mode, void *_prefs, int
 	struct ast_format_list format;
 	int frames_per_packet;
 	int max_frames_per_packet;
+	H323Capability *cap;
 
 	localCapabilities.RemoveAll();
 
 	if (h323debug) {
-		cout << "Setting capabilities to " << ast_getformatname_multiple(caps_str, sizeof(caps_str), cap) << endl;
+		cout << "Setting capabilities to " << ast_getformatname_multiple(caps_str, sizeof(caps_str), caps) << endl;
 		ast_codec_pref_string(prefs, caps_str, sizeof(caps_str));
 		cout << "Capabilities in preference order is " << caps_str << endl;
 	}
@@ -1449,7 +1469,7 @@ void MyH323Connection::SetCapabilities(int cap, int dtmf_mode, void *_prefs, int
 				y <<= 1;
 			codec = y;
 		}
-		if (!(cap & codec) || (alreadysent & codec) || !(codec & AST_FORMAT_AUDIO_MASK))
+		if (!(caps & codec) || (alreadysent & codec) || !(codec & AST_FORMAT_AUDIO_MASK))
 			continue;
 		alreadysent |= codec;
 		format = ast_codec_pref_getsize(prefs, codec);
@@ -1518,23 +1538,64 @@ void MyH323Connection::SetCapabilities(int cap, int dtmf_mode, void *_prefs, int
 		}
 	}
 
-	lastcap++;
-	lastcap = localCapabilities.SetCapability(0, lastcap, new H323_UserInputCapability(H323_UserInputCapability::HookFlashH245));
+	cap = new H323_UserInputCapability(H323_UserInputCapability::HookFlashH245);
+	if (cap && cap->IsUsable(*this)) {
+		lastcap++;
+		lastcap = localCapabilities.SetCapability(0, lastcap, cap);
+	} else if (cap)
+		delete cap;				/* Capability is not usable */
 
-	lastcap++;
 	dtmfMode = dtmf_mode;
-	if ((dtmfMode & H323_DTMF_INBAND)) {
-		localCapabilities.SetCapability(0, lastcap, new H323_UserInputCapability(H323_UserInputCapability::BasicString));
-		sendUserInputMode = SendUserInputAsString;
-	} else {
-		lastcap = localCapabilities.SetCapability(0, lastcap, new H323_UserInputCapability(H323_UserInputCapability::SignalToneRFC2833));
-		/* Cisco sends DTMF only through h245-alphanumeric or h245-signal, no support for RFC2833 */
-		lastcap = localCapabilities.SetCapability(0, lastcap, new H323_UserInputCapability(H323_UserInputCapability::SignalToneH245));
-		sendUserInputMode = SendUserInputAsTone;	/* RFC2833 transmission handled at Asterisk level */
+	if (h323debug) {
+		cout << "DTMF mode is " << (int)dtmfMode << endl;
+	}
+	if (dtmfMode) {
+		lastcap++;
+		if (dtmfMode == H323_DTMF_INBAND) {
+			cap = new H323_UserInputCapability(H323_UserInputCapability::BasicString);
+			if (cap && cap->IsUsable(*this)) {
+				lastcap = localCapabilities.SetCapability(0, lastcap, cap);
+			} else if (cap)
+				delete cap;		/* Capability is not usable */
+			sendUserInputMode = SendUserInputAsString;
+		} else {
+			if ((dtmfMode & H323_DTMF_RFC2833) != 0) {
+				cap = new H323_UserInputCapability(H323_UserInputCapability::SignalToneRFC2833);
+				if (cap && cap->IsUsable(*this))
+					lastcap = localCapabilities.SetCapability(0, lastcap, cap);
+				else {
+					dtmfMode |= H323_DTMF_SIGNAL;
+					if (cap)
+						delete cap;	/* Capability is not usable */
+				}
+			}
+			if ((dtmfMode & H323_DTMF_CISCO) != 0) {
+				/* Try Cisco's RTP DTMF relay too, but prefer RFC2833 or h245-signal */
+				cap = new AST_CiscoDtmfCapability();
+				if (cap && cap->IsUsable(*this)) {
+					lastcap = localCapabilities.SetCapability(0, lastcap, cap);
+					/* We cannot send Cisco RTP DTMFs, use h245-signal instead */
+					dtmfMode |= H323_DTMF_SIGNAL;
+				} else {
+					dtmfMode |= H323_DTMF_SIGNAL;
+					if (cap)
+						delete cap;	/* Capability is not usable */
+				}
+			}
+			if ((dtmfMode & H323_DTMF_SIGNAL) != 0) {
+				/* Cisco usually sends DTMF correctly only through h245-alphanumeric or h245-signal */
+				cap = new H323_UserInputCapability(H323_UserInputCapability::SignalToneH245);
+				if (cap && cap->IsUsable(*this))
+					lastcap = localCapabilities.SetCapability(0, lastcap, cap);
+				else if (cap)
+					delete cap;	/* Capability is not usable */
+			}
+			sendUserInputMode = SendUserInputAsTone;	/* RFC2833 transmission handled at Asterisk level */
+		}
 	}
 
 	if (h323debug) {
-		cout << "Allowed Codecs:\n\t" << setprecision(2) << localCapabilities << endl;
+		cout << "Allowed Codecs for " << GetCallToken() << " (" << GetSignallingChannel()->GetLocalAddress() << "):\n\t" << setprecision(2) << localCapabilities << endl;
 	}
 }
 
diff --git a/channels/h323/ast_h323.h b/channels/h323/ast_h323.h
index 4c85c2442..7199b0f34 100644
--- a/channels/h323/ast_h323.h
+++ b/channels/h323/ast_h323.h
@@ -112,7 +112,7 @@ public:
 	unsigned progressAlert;
 	int cause;
 
-	RTP_DataFrame::PayloadTypes dtmfCodec;
+	RTP_DataFrame::PayloadTypes dtmfCodec[2];
 	int dtmfMode;
 };
 
diff --git a/channels/h323/caps_h323.cxx b/channels/h323/caps_h323.cxx
index 1a14e2337..ebb90f3f2 100644
--- a/channels/h323/caps_h323.cxx
+++ b/channels/h323/caps_h323.cxx
@@ -18,6 +18,7 @@ H323_REGISTER_CAPABILITY(AST_G729Capability,  OPAL_G729);
 H323_REGISTER_CAPABILITY(AST_G729ACapability, OPAL_G729A);
 H323_REGISTER_CAPABILITY(AST_GSM0610Capability, OPAL_GSM0610);
 H323_REGISTER_CAPABILITY(AST_CiscoG726Capability, CISCO_G726r32);
+H323_REGISTER_CAPABILITY(AST_CiscoDtmfCapability, CISCO_DTMF_RELAY);
 
 OPAL_MEDIA_FORMAT_DECLARE(OpalG711ALaw64kFormat,
 	OPAL_G711_ALAW_64K,
@@ -99,6 +100,18 @@ OPAL_MEDIA_FORMAT_DECLARE(OpalCiscoG726Format,
 	8,		// 1 millisecond
 	OpalMediaFormat::AudioTimeUnits,
 	0);
+#if 0
+OPAL_MEDIA_FORMAT_DECLARE(OpalCiscoDTMFRelayFormat,
+	CISCO_DTMF_RELAY,
+	OpalMediaFormat::DefaultAudioSessionID,
+	(RTP_DataFrame::PayloadTypes)121, // Choose this for Cisco IOS compatibility
+	TRUE,	// Needs jitter
+	100,	// bits/sec
+	4,		// bytes/frame
+	8*150,	// 150 millisecond
+	OpalMediaFormat::AudioTimeUnits,
+	0);
+#endif
 
 /*
  * Capability: G.711
diff --git a/channels/h323/chan_h323.h b/channels/h323/chan_h323.h
index 8905e4eea..13296ed1a 100644
--- a/channels/h323/chan_h323.h
+++ b/channels/h323/chan_h323.h
@@ -52,7 +52,7 @@ typedef struct call_options {
 	int				progress_setup;
 	int				progress_alert;
 	int				progress_audio;
-	int				dtmfcodec;
+	int				dtmfcodec[2];
 	int				dtmfmode;
 	int				capability;
 	int				bridge;
@@ -172,7 +172,7 @@ extern answer_call_cb on_answer_call;
 /* This is a callback prototype function, called when
    we know which RTP payload type RFC2833 will be
    transmitted */
-typedef void (*rfc2833_cb)(unsigned, const char *, int);
+typedef void (*rfc2833_cb)(unsigned, const char *, int, int);
 extern rfc2833_cb on_set_rfc2833_payload;
 
 typedef void (*hangup_cb)(unsigned, const char *, int);
@@ -188,7 +188,9 @@ extern setpeercapabilities_cb on_setpeercapabilities;
 extern int h323debug;
 
 #define H323_DTMF_RFC2833	(1 << 0)
-#define H323_DTMF_INBAND	(1 << 1)
+#define H323_DTMF_CISCO		(1 << 1)
+#define H323_DTMF_SIGNAL	(1 << 2)
+#define H323_DTMF_INBAND	(1 << 3)
 
 #define H323_DTMF_RFC2833_PT	101
 #define H323_DTMF_CISCO_PT		121