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
This commit is contained in:
Torrey Searle
2017-06-15 10:12:41 +02:00
committed by George Joseph
parent 45df25a579
commit fb7247c57c
7 changed files with 115 additions and 12 deletions

View File

@@ -207,7 +207,7 @@ 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 || 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) {
@@ -230,7 +230,7 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
}
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;
@@ -296,6 +296,16 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
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));
@@ -404,7 +414,8 @@ static int set_caps(struct ast_sip_session *session,
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);
@@ -1136,7 +1147,7 @@ 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 || 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);