res_pjsip: Add an 'auto' option for DTMF Mode

This patch adds support for automatically detecting the type of DTMF that a
PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto',
the channel created for an endpoint will attempt to determine if RFC 4733
DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type
for the channel will be set to inband.

Review: https://reviewboard.asterisk.org/r/4438

ASTERISK-24706 #close
Reported by: yaron nahum
patches:
  yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676)
........

Merged revisions 434637 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434638 65c4cc65-6c06-0410-ace0-fbb531ad65f3

4 files changed, 35 insertions, 7 deletions

diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 0a154f4ff..fcd8516b6 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -210,6 +210,10 @@
 							<enum name="info">
 								<para>DTMF is sent as SIP INFO packets.</para>
 							</enum>
+                                                        <enum name="auto">
+                                                                <para>DTMF is sent as RFC 4733 if the other side supports it or as INBAND if not.</para>
+                                                        </enum>
+
 						</enumlist>
 					</description>
 				</configOption>
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index eacbef064..0eecb5e0a 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -109,6 +109,8 @@ static int dtmf_handler(const struct aco_option *opt, struct ast_variable *var,
 		endpoint->dtmf = AST_SIP_DTMF_INBAND;
 	} else if (!strcasecmp(var->value, "info")) {
 		endpoint->dtmf = AST_SIP_DTMF_INFO;
+        } else if (!strcasecmp(var->value, "auto")) {
+                endpoint->dtmf = AST_SIP_DTMF_AUTO;
 	} else if (!strcasecmp(var->value, "none")) {
 		endpoint->dtmf = AST_SIP_DTMF_NONE;
 	} else {
@@ -129,6 +131,8 @@ static int dtmf_to_str(const void *obj, const intptr_t *args, char **buf)
 		*buf = "inband"; break;
 	case AST_SIP_DTMF_INFO :
 		*buf = "info"; break;
+       case AST_SIP_DTMF_AUTO :
+                *buf = "auto"; break;
 	default:
 		*buf = "none";
 	}
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index f396bfea0..2ecd11a84 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -50,6 +50,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #include "asterisk/sched.h"
 #include "asterisk/acl.h"
 #include "asterisk/sdp_srtp.h"
+#include "asterisk/dsp.h"
 
 #include "asterisk/res_pjsip.h"
 #include "asterisk/res_pjsip_session.h"
@@ -123,7 +124,7 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
 		ice->stop(session_media->rtp);
 	}
 
-	if (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733) {
+	if (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733 || session->endpoint->dtmf == AST_SIP_DTMF_AUTO) {
 		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) {
@@ -143,13 +144,14 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
 	return 0;
 }
 
-static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp_media *stream, struct ast_rtp_codecs *codecs)
+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)
 {
 	pjmedia_sdp_attr *attr;
 	pjmedia_sdp_rtpmap *rtpmap;
 	pjmedia_sdp_fmtp fmtp;
 	struct ast_format *format;
-	int i, num = 0;
+	int i, num = 0, tel_event = 0;
 	char name[256];
 	char media[20];
 	char fmt_param[256];
@@ -171,6 +173,9 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
 		}
 
 		ast_copy_pj_str(name, &rtpmap->enc_name, sizeof(name));
+                if (strcmp(name,"telephone-event") == 0) {
+                        tel_event++;
+                }
 		ast_copy_pj_str(media, (pj_str_t*)&stream->desc.media, sizeof(media));
 		ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, NULL, pj_strtoul(&stream->desc.fmt[i]),
 							     media, name, 0, rtpmap->clock_rate);
@@ -200,7 +205,9 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
 			}
 		}
 	}
-
+	if ((tel_event==0) && (session->endpoint->dtmf == AST_SIP_DTMF_AUTO)) {
+                ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
+	}
 	/* 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));
@@ -221,6 +228,7 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
 	int fmts = 0;
 	int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
 		ast_format_cap_count(session->direct_media_cap);
+	int dsp_features = 0;
 
 	if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
 	    !(peer = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
@@ -238,7 +246,7 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
 	}
 
 	/* get the capabilities on the peer */
-	get_codecs(session, stream, &codecs);
+	get_codecs(session, stream, &codecs,  session_media);
 	ast_rtp_codecs_payload_formats(&codecs, peer, &fmts);
 
 	/* get the joint capabilities between peer and endpoint */
@@ -288,6 +296,18 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi
 		ast_channel_nativeformats_set(session->channel, caps);
 		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)
+		    && (ast_rtp_instance_dtmf_mode_get(session_media->rtp) == AST_RTP_DTMF_MODE_RFC2833)
+		    && (session->dsp)) {
+			dsp_features = ast_dsp_get_features(session->dsp);
+			dsp_features &= ~DSP_FEATURE_DIGIT_DETECT;
+			if (dsp_features) {
+				ast_dsp_set_features(session->dsp, dsp_features);
+			} else {
+				ast_dsp_free(session->dsp);
+				session->dsp = NULL;
+			}
+		}
 		ast_channel_unlock(session->channel);
 
 		ao2_ref(fmt, -1);
@@ -952,7 +972,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
 	pj_str_t stmp;
 	pjmedia_sdp_attr *attr;
 	int index = 0;
-	int noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733) ? AST_RTP_DTMF : 0;
+	int noncodec = (session->endpoint->dtmf == AST_SIP_DTMF_RFC_4733 || session->endpoint->dtmf == AST_SIP_DTMF_AUTO) ? 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);
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 4854c51d0..87ce2b0c1 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -1283,7 +1283,7 @@ struct ast_sip_session *ast_sip_session_alloc(struct ast_sip_endpoint *endpoint,
 	session->inv_session = inv_session;
 	session->req_caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
 
-	if (endpoint->dtmf == AST_SIP_DTMF_INBAND) {
+	if ((endpoint->dtmf == AST_SIP_DTMF_INBAND) || (endpoint->dtmf == AST_SIP_DTMF_AUTO)) {
 		dsp_features |= DSP_FEATURE_DIGIT_DETECT;
 	}