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res_pjsip: Add an 'auto' option for DTMF Mode

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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
This commit is contained in:
Matthew Jordan
2015-04-10 17:56:47 +00:00
parent f69e46de25
commit 8bae18ab93
9 changed files with 128 additions and 8 deletions
Download Patch File Download Diff File Expand all files Collapse all files

32
res/res_pjsip_sdp_rtp.c
View File

@@ -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);