[asterisk-commits] res pjsip: Add DTMF INFO Failback mode (asterisk[13])

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Mon Jun 26 18:06:19 CDT 2017


Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/5842 )

Change subject: res_pjsip:  Add DTMF INFO Failback mode
......................................................................

res_pjsip:  Add DTMF INFO Failback mode

The existing auto dtmf mode reverts to inband if 4733 fails to be
negotiated.  This patch adds a new mode auto_info which will
switch to INFO instead of inband if 4733 is not available.

ASTERISK-27066 #close

Change-Id: Id185b11e84afd9191a2f269e8443019047765e91
---
M CHANGES
M channels/chan_pjsip.c
A contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py
M include/asterisk/res_pjsip.h
M res/res_pjsip.c
M res/res_pjsip/pjsip_configuration.c
M res/res_pjsip_sdp_rtp.c
7 files changed, 121 insertions(+), 18 deletions(-)

Approvals:
  Joshua Colp: Looks good to me, but someone else must approve
  Matthew Fredrickson: Looks good to me, approved
  Jenkins2: Approved for Submit



diff --git a/CHANGES b/CHANGES
index 1b87dbf..c2a9c8f 100644
--- a/CHANGES
+++ b/CHANGES
@@ -25,6 +25,10 @@
    whether to notify dialog-info state 'early' or 'confirmed' on Ringing
    when already INUSE.
 
+ * The endpoint option 'dtmf_mode' has a new option 'auto_dtmf' added. This
+   mode works similar to 'auto' except uses DTMF INFO as fallback instead of
+   INBAND.
+
 res_agi
 ------------------
  * The EAGI() application will now look for a dialplan variable named
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 486a237..d1691b8 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -1708,14 +1708,20 @@
 		}
 
 		ast_rtp_instance_dtmf_begin(media->rtp, digit);
-                break;
+		break;
 	case AST_SIP_DTMF_AUTO:
-                       if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
-                        return -1;
-                }
+		if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
+			return -1;
+		}
 
-                ast_rtp_instance_dtmf_begin(media->rtp, digit);
-                break;
+		ast_rtp_instance_dtmf_begin(media->rtp, digit);
+		break;
+	case AST_SIP_DTMF_AUTO_INFO:
+		if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_NONE)) {
+			return -1;
+		}
+		ast_rtp_instance_dtmf_begin(media->rtp, digit);
+		break;
 	case AST_SIP_DTMF_NONE:
 		break;
 	case AST_SIP_DTMF_INBAND:
@@ -1816,6 +1822,20 @@
 	int res = 0;
 
 	switch (channel->session->endpoint->dtmf) {
+	case AST_SIP_DTMF_AUTO_INFO:
+	{
+		if (!media || !media->rtp) {
+			return -1;
+		}
+		if (ast_rtp_instance_dtmf_mode_get(media->rtp) != AST_RTP_DTMF_MODE_NONE) {
+			ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 negotiated so using it.\n", ast_channel_name(ast));
+			ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
+			break;
+		}
+		/* If RFC_4733 was not negotiated, fail through to the DTMF_INFO processing */
+		ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 NOT negotiated using INFO instead.\n", ast_channel_name(ast));
+	}
+
 	case AST_SIP_DTMF_INFO:
 	{
 		struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
@@ -1848,14 +1868,15 @@
 		}
 
 		ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
-                break;
-        case AST_SIP_DTMF_AUTO:
-                if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
-                        return -1;
-                }
+		break;
+	case AST_SIP_DTMF_AUTO:
+		if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
+			 return -1;
+		}
 
-                ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
-                break;
+		ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
+		break;
+
 
 	case AST_SIP_DTMF_NONE:
 		break;
diff --git a/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py b/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py
new file mode 100644
index 0000000..dbc8ce9
--- /dev/null
+++ b/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py
@@ -0,0 +1,57 @@
+"""Add auto_info to endpoint dtmf_mode
+
+Revision ID: 164abbd708c
+Revises: 86bb1efa278d
+Create Date: 2017-06-19 13:55:15.354706
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '164abbd708c'
+down_revision = '86bb1efa278d'
+
+from alembic import op
+import sqlalchemy as sa
+
+OLD_ENUM = ['rfc4733', 'inband', 'info', 'auto']
+NEW_ENUM = ['rfc4733', 'inband', 'info', 'auto', 'auto_info']
+
+old_type = sa.Enum(*OLD_ENUM, name='pjsip_dtmf_mode_values_v2')
+new_type = sa.Enum(*NEW_ENUM, name='pjsip_dtmf_mode_values_v3')
+
+def upgrade():
+    context = op.get_context()
+
+    # Upgrading to this revision WILL clear your directmedia values.
+    if context.bind.dialect.name != 'postgresql':
+        op.alter_column('ps_endpoints', 'dtmf_mode',
+                        type_=new_type,
+                        existing_type=old_type)
+    else:
+        enum = ENUM('rfc4733', 'inband', 'info', 'auto', 'auto_info',
+                    name='pjsip_dtmf_mode_values_v3')
+        enum.create(op.get_bind(), checkfirst=False)
+
+        op.execute('ALTER TABLE ps_endpoints ALTER COLUMN dtmf_mode TYPE'
+                   ' pjsip_dtmf_mode_values_v3 USING'
+                   ' dtmf_mode::text::pjsip_dtmf_mode_values_v3')
+
+        ENUM(name="pjsip_dtmf_mode_values_v2").drop(op.get_bind(), checkfirst=False)
+
+def downgrade():
+    context = op.get_context()
+
+    if context.bind.dialect.name != 'postgresql':
+        op.alter_column('ps_endpoints', 'dtmf_mode',
+                        type_=old_type,
+                        existing_type=new_type)
+    else:
+        enum = ENUM('rfc4733', 'inband', 'info', 'auto',
+                    name='pjsip_dtmf_mode_values_v2')
+        enum.create(op.get_bind(), checkfirst=False)
+
+        op.execute('ALTER TABLE ps_endpoints ALTER COLUMN dtmf_mode TYPE'
+                   ' pjsip_dtmf_mode_values USING'
+                   ' dtmf_mode::text::pjsip_dtmf_mode_values_v2')
+
+        ENUM(name="pjsip_dtmf_mode_values_v3").drop(op.get_bind(), checkfirst=False)
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 04e0c65..b5a0288 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -363,6 +363,8 @@
 	AST_SIP_DTMF_INFO,
 	/*! Use SIP 4733 if supported by the other side or INBAND if not */
 	AST_SIP_DTMF_AUTO,
+	/*! Use SIP 4733 if supported by the other side or INFO DTMF (blech) if not */
+	AST_SIP_DTMF_AUTO_INFO,
 };
 
 /*!
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index f79ebfd..057ae33 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -229,6 +229,9 @@
 							<enum name="auto">
 								<para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</para>
 							</enum>
+							<enum name="auto_info">
+								<para>DTMF is sent as RFC 4733 if the other side supports it or as SIP INFO if not.</para>
+							</enum>
 						</enumlist>
 					</description>
 				</configOption>
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index d2d1a0c..9604ff2 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -373,6 +373,8 @@
 		endpoint->dtmf = AST_SIP_DTMF_RFC_4733;
 	} else if (!strcasecmp(var->value, "inband")) {
 		endpoint->dtmf = AST_SIP_DTMF_INBAND;
+	} else if (!strcasecmp(var->value, "auto_info")) {
+		endpoint->dtmf = AST_SIP_DTMF_AUTO_INFO;
 	} else if (!strcasecmp(var->value, "info")) {
 		endpoint->dtmf = AST_SIP_DTMF_INFO;
 	} else if (!strcasecmp(var->value, "auto")) {
@@ -397,8 +399,11 @@
 		*buf = "inband"; break;
 	case AST_SIP_DTMF_INFO :
 		*buf = "info"; break;
-       case AST_SIP_DTMF_AUTO :
+	case AST_SIP_DTMF_AUTO :
 		*buf = "auto"; break;
+	case AST_SIP_DTMF_AUTO_INFO :
+		*buf = "auto_info";
+		break;
 	default:
 		*buf = "none";
 	}
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index d39842f..a6bd2d7 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -246,7 +246,7 @@
 		ice->stop(session_media->rtp);
 	}
 
-	if (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733 || session->endpoint->dtmf == AST_SIP_DTMF_AUTO) {
+	if (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733 || session->endpoint->dtmf == AST_SIP_DTMF_AUTO || session->endpoint->dtmf == AST_SIP_DTMF_AUTO_INFO) {
 		ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
 		ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 1);
 	} else if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
@@ -269,7 +269,7 @@
 }
 
 static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp_media *stream, struct ast_rtp_codecs *codecs,
-       struct ast_sip_session_media *session_media)
+	struct ast_sip_session_media *session_media)
 {
 	pjmedia_sdp_attr *attr;
 	pjmedia_sdp_rtpmap *rtpmap;
@@ -335,6 +335,16 @@
 	if (!tel_event && (session->endpoint->dtmf == AST_SIP_DTMF_AUTO)) {
 		ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
 	}
+
+	if (session->endpoint->dtmf == AST_SIP_DTMF_AUTO_INFO) {
+		if  (tel_event) {
+			ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
+		} else {
+			ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_NONE);
+		}
+	}
+
+
 	/* Get the packetization, if it exists */
 	if ((attr = pjmedia_sdp_media_find_attr2(stream, "ptime", NULL))) {
 		unsigned long framing = pj_strtoul(pj_strltrim(&attr->value));
@@ -429,7 +439,8 @@
 			ast_set_read_format(session->channel, ast_channel_readformat(session->channel));
 			ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
 		}
-		if ((session->endpoint->dtmf == AST_SIP_DTMF_AUTO)
+
+		if ( ((session->endpoint->dtmf == AST_SIP_DTMF_AUTO) || (session->endpoint->dtmf == AST_SIP_DTMF_AUTO_INFO) )
 		    && (ast_rtp_instance_dtmf_mode_get(session_media->rtp) == AST_RTP_DTMF_MODE_RFC2833)
 		    && (session->dsp)) {
 			dsp_features = ast_dsp_get_features(session->dsp);
@@ -1149,7 +1160,7 @@
 	pj_str_t stmp;
 	pjmedia_sdp_attr *attr;
 	int index = 0;
-	int noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733 || session->endpoint->dtmf == AST_SIP_DTMF_AUTO) ? AST_RTP_DTMF : 0;
+	int noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733 || session->endpoint->dtmf == AST_SIP_DTMF_AUTO || session->endpoint->dtmf == AST_SIP_DTMF_AUTO_INFO) ? AST_RTP_DTMF : 0;
 	int min_packet_size = 0, max_packet_size = 0;
 	int rtp_code;
 	RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);

-- 
To view, visit https://gerrit.asterisk.org/5842
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-MessageType: merged
Gerrit-Change-Id: Id185b11e84afd9191a2f269e8443019047765e91
Gerrit-Change-Number: 5842
Gerrit-PatchSet: 4
Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
Gerrit-Reviewer: Alexei Gradinari <alex2grad at gmail.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matthew Fredrickson <creslin at digium.com>
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