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

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 897aa2d498e575d2041c3baa6e1886ab444ed335..745347cb18384f5fdb2cd0284950240503db0c3f 100644 (file)
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -548,7 +548,7 @@ static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *se
        int exists;
 
        /* If we only needed this DSP for fax detection purposes we can just drop it now */
-       if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
+       if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND || session->endpoint->dtmf == AST_SIP_DTMF_AUTO) {
                ast_dsp_set_features(session->dsp, DSP_FEATURE_DIGIT_DETECT);
        } else {
                ast_dsp_free(session->dsp);
@@ -1522,6 +1522,14 @@ static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
                }
 
                ast_rtp_instance_dtmf_begin(media->rtp, digit);
+                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_begin(media->rtp, digit);
+                break;
        case AST_SIP_DTMF_NONE:
                break;
        case AST_SIP_DTMF_INBAND:
@@ -1625,6 +1633,15 @@ static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned in
                }
 
                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;
+                }
+
+                ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
+                break;
+
        case AST_SIP_DTMF_NONE:
                break;
        case AST_SIP_DTMF_INBAND:

diff --git a/contrib/ast-db-manage/config/versions/31cd4f4891ec_add_auto_dtmf_mode.py b/contrib/ast-db-manage/config/versions/31cd4f4891ec_add_auto_dtmf_mode.py
new file mode 100644 (file)
index 0000000..fd1b6c7
--- /dev/null
+++ b/contrib/ast-db-manage/config/versions/31cd4f4891ec_add_auto_dtmf_mode.py
@@ -0,0 +1,63 @@
+"""Add auto DTMF mode
+
+Revision ID: 31cd4f4891ec
+Revises: 23530d604b96
+Create Date: 2015-04-10 12:36:51.619419
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '31cd4f4891ec'
+down_revision = '23530d604b96'
+
+from alembic import op
+import sqlalchemy as sa
+
+OLD_ENUM = ['rfc4733', 'inband', 'info']
+NEW_ENUM = ['rfc4733', 'inband', 'info', 'auto']
+
+old_type = sa.Enum(*OLD_ENUM, name='pjsip_dtmf_mode_values')
+new_type = sa.Enum(*NEW_ENUM, name='pjsip_dtmf_mode_values_v2')
+
+tcr = sa.sql.table('ps_endpoints', sa.Column('dtmf_mode', new_type,
+                   nullable=True))
+
+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',
+                    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_v2 USING'
+                   ' dtmf_mode::text::pjsip_dtmf_mode_values_v2')
+
+        ENUM(name="pjsip_dtmf_mode_values").drop(op.get_bind(), checkfirst=False)
+
+def downgrade():
+    context = op.get_context()
+
+    op.execute(tcr.update().where(tcr.c.directmedia==u'outgoing')
+               .values(directmedia=None))
+
+    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',
+                    name='pjsip_dtmf_mode_values')
+        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')
+
+        ENUM(name="pjsip_dtmf_mode_values_v2").drop(op.get_bind(), checkfirst=False)

diff --git a/include/asterisk/dsp.h b/include/asterisk/dsp.h
index 16262c05d468dd49c6d95ea36dcd9fb6911885ad..7e84ebe9caa11bc41a0e8b43836fae11f6e408c6 100644 (file)
--- a/include/asterisk/dsp.h
+++ b/include/asterisk/dsp.h
@@ -138,6 +138,9 @@ void ast_dsp_digitreset(struct ast_dsp *dsp);
 /*! \brief Select feature set */
 void ast_dsp_set_features(struct ast_dsp *dsp, int features);
 
+/*! \brief Get features */
+int ast_dsp_get_features(struct ast_dsp *dsp);
+
 /*! \brief Get pending DTMF/MF digits */
 int ast_dsp_getdigits(struct ast_dsp *dsp, char *buf, int max);
 

diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index 6e3a9ea930a7726f1257ed4c7404f3621b32daae..2358a72813dc654c1be4beaced74355c3757063f 100644 (file)
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -252,6 +252,8 @@ enum ast_sip_dtmf_mode {
        AST_SIP_DTMF_INBAND,
        /*! Use SIP INFO DTMF (blech) */
        AST_SIP_DTMF_INFO,
+       /*! Use SIP 4733 if supported by the other side or INBAND if not */
+       AST_SIP_DTMF_AUTO,
 };
 
 /*!

diff --git a/main/dsp.c b/main/dsp.c
index 335fb3db0ff76ca9ff78da4c33a2d504c768362f..087416358e3609789a651d3c68589675e4eff8ef 100644 (file)
--- a/main/dsp.c
+++ b/main/dsp.c
@@ -1702,6 +1702,13 @@ void ast_dsp_set_features(struct ast_dsp *dsp, int features)
        }
 }
 
+
+int ast_dsp_get_features(struct ast_dsp *dsp)
+{
+        return (dsp->features);
+}
+
+
 void ast_dsp_free(struct ast_dsp *dsp)
 {
        ast_free(dsp);

diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 0a154f4ff01f940c36ed8ec4aacb86ec42f38321..fcd8516b654dd5114d8b54f8546c8866bfa6fb83 100644 (file)
--- 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 eacbef0641fa0cc2b6999bb733ac51811e11b725..0eecb5e0a22fc1c2caff910cef94ed3ae714c331 100644 (file)
--- 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 f396bfea085a0889c17382425580a0904d3046cf..2ecd11a84404f3363914938bef7cd7c505ffc804 100644 (file)
--- 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 4854c51d03a0f8e51935e54ad93904e07e1a8290..87ce2b0c1b31b5d43b9107bf8b2614438865ce4b 100644 (file)
--- 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;
        }