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chan_pjsip: add a new function PJSIP_DTMF_MODE

Browse Source 
This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis

ASTERISK-27085 #close

Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612
14
Torrey Searle 9 years ago committed by George Joseph
parent 6b7471df28
commit aa6d5c83f1
10 changed files with 293 additions and 37 deletions
Show Stats Download Patch File Download Diff File

3
CHANGES
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@ -33,6 +33,9 @@ res_musiconhold
which sends signals to the application and its descendants directly, or
"process" which sends signals only to the application itself.
* New dialplan function PJSIP_DTMF_MODE added to get or change the DTMF mode
of a channel on a per-call basis.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.5.0 to Asterisk 14.6.0 ------------
------------------------------------------------------------------------------

17
channels/chan_pjsip.c
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@ -1750,7 +1750,7 @@ static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
int res = 0;
switch (channel->session->endpoint->dtmf) {
switch (channel->session->dtmf) {
case AST_SIP_DTMF_RFC_4733:
if (!media || !media->rtp) {
return -1;
@ -1870,7 +1870,7 @@ static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned in
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
int res = 0;
switch (channel->session->endpoint->dtmf) {
switch (channel->session->dtmf) {
case AST_SIP_DTMF_AUTO_INFO:
{
if (!media || !media->rtp) {
@ -2682,6 +2682,12 @@ static struct ast_custom_function media_offer_function = {
.write = pjsip_acf_media_offer_write
};
static struct ast_custom_function dtmf_mode_function = {
.name = "PJSIP_DTMF_MODE",
.read = pjsip_acf_dtmf_mode_read,
.write = pjsip_acf_dtmf_mode_write
};
static struct ast_custom_function session_refresh_function = {
.name = "PJSIP_SEND_SESSION_REFRESH",
.write = pjsip_acf_session_refresh_write,
@ -2726,6 +2732,11 @@ static int load_module(void)
goto end;
}
if (ast_custom_function_register(&dtmf_mode_function)) {
ast_log(LOG_WARNING, "Unable to register PJSIP_DTMF_MODE dialplan function\n");
goto end;
}
if (ast_custom_function_register(&session_refresh_function)) {
ast_log(LOG_WARNING, "Unable to register PJSIP_SEND_SESSION_REFRESH dialplan function\n");
goto end;
@ -2785,6 +2796,7 @@ static int load_module(void)
end:
ao2_cleanup(pjsip_uids_onhold);
pjsip_uids_onhold = NULL;
ast_custom_function_unregister(&dtmf_mode_function);
ast_custom_function_unregister(&media_offer_function);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
ast_custom_function_unregister(&session_refresh_function);
@ -2807,6 +2819,7 @@ static int unload_module(void)
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
ast_sip_session_unregister_supplement(&call_pickup_supplement);
ast_custom_function_unregister(&dtmf_mode_function);
ast_custom_function_unregister(&media_offer_function);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
ast_custom_function_unregister(&session_refresh_function);

152
channels/pjsip/dialplan_functions.c
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@ -68,6 +68,18 @@
<ref type="function">PJSIP_SEND_SESSION_REFRESH</ref>
</see-also>
</function>
<function name="PJSIP_DTMF_MODE" language="en_US">
<synopsis>
Get or change the DTMF mode for a SIP call.
</synopsis>
<syntax>
</syntax>
<description>
<para>When read, returns the current DTMF mode</para>
<para>When written, sets the current DTMF mode</para>
<para>This function uses the same DTMF mode naming as the dtmf_mode configuration option</para>
</description>
</function>
<function name="PJSIP_SEND_SESSION_REFRESH" language="en_US">
<synopsis>
W/O: Initiate a session refresh via an UPDATE or re-INVITE on an established media session
@ -440,6 +452,7 @@ ASTERISK_REGISTER_FILE()
#include "asterisk/app.h"
#include "asterisk/channel.h"
#include "asterisk/format.h"
#include "asterisk/dsp.h"
#include "asterisk/pbx.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
@ -1039,6 +1052,34 @@ int pjsip_acf_media_offer_write(struct ast_channel *chan, const char *cmd, char
return ast_sip_push_task_synchronous(channel->session->serializer, media_offer_write_av, &mdata);
}
int pjsip_acf_dtmf_mode_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct ast_sip_channel_pvt *channel;
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
ast_channel_lock(chan);
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
ast_channel_unlock(chan);
return -1;
}
channel = ast_channel_tech_pvt(chan);
if (ast_sip_dtmf_to_str(channel->session->dtmf, buf, len) < 0) {
ast_log(LOG_WARNING, "Unknown DTMF mode %d on PJSIP channel %s\n", channel->session->dtmf, ast_channel_name(chan));
ast_channel_unlock(chan);
return -1;
}
ast_channel_unlock(chan);
return 0;
}
struct refresh_data {
struct ast_sip_session *session;
enum ast_sip_session_refresh_method method;
@ -1067,6 +1108,117 @@ static int sip_session_response_cb(struct ast_sip_session *session, pjsip_rx_dat
return 0;
}
static int dtmf_mode_refresh_cb(void *obj)
{
struct refresh_data *data = obj;
if (data->session->inv_session->state == PJSIP_INV_STATE_CONFIRMED) {
ast_debug(3, "Changing DTMF mode on channel %s after OFFER/ANSER completion. Sending session refresh\n", ast_channel_name(data->session->channel));
ast_sip_session_refresh(data->session, NULL, NULL,
sip_session_response_cb, data->method, 1);
}
return 0;
}
int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct ast_sip_channel_pvt *channel;
struct chan_pjsip_pvt *pjsip_pvt;
int dsp_features = 0;
int dtmf = -1;
struct refresh_data rdata = {
.method = AST_SIP_SESSION_REFRESH_METHOD_INVITE,
};
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
ast_channel_lock(chan);
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
ast_channel_unlock(chan);
return -1;
}
channel = ast_channel_tech_pvt(chan);
rdata.session = channel->session;
dtmf = ast_sip_str_to_dtmf(value);
if (dtmf == -1) {
ast_log(LOG_WARNING, "Cannot set DTMF mode to '%s' on channel '%s' as value is invalid.\n", value,
ast_channel_name(chan));
ast_channel_unlock(chan);
return -1;
}
if (channel->session->dtmf == dtmf) {
/* DTMF mode unchanged, nothing to do! */
ast_channel_unlock(chan);
return 0;
}
channel->session->dtmf = dtmf;
pjsip_pvt = channel->pvt;
if (pjsip_pvt->media[SIP_MEDIA_AUDIO] && (pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp) {
if (channel->session->dtmf == AST_SIP_DTMF_RFC_4733) {
ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 1);
ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_RFC2833);
} else if (channel->session->dtmf == AST_SIP_DTMF_INFO) {
ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_NONE);
} else if (channel->session->dtmf == AST_SIP_DTMF_INBAND) {
ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_INBAND);
} else if (channel->session->dtmf == AST_SIP_DTMF_NONE) {
ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_NONE);
} else if (channel->session->dtmf == AST_SIP_DTMF_AUTO) {
if (ast_rtp_instance_dtmf_mode_get((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp) != AST_RTP_DTMF_MODE_RFC2833) {
/* no RFC4733 negotiated, enable inband */
ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_INBAND);
}
} else if (channel->session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
ast_rtp_instance_set_prop((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_PROPERTY_DTMF, 0);
if (ast_rtp_instance_dtmf_mode_get((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp) == AST_RTP_DTMF_MODE_INBAND) {
/* if inband, switch to INFO */
ast_rtp_instance_dtmf_mode_set((pjsip_pvt->media[SIP_MEDIA_AUDIO])->rtp, AST_RTP_DTMF_MODE_NONE);
}
}
}
if (channel->session->dsp) {
dsp_features = ast_dsp_get_features(channel->session->dsp);
}
if (channel->session->dtmf == AST_SIP_DTMF_INBAND ||
channel->session->dtmf == AST_SIP_DTMF_AUTO) {
dsp_features |= DSP_FEATURE_DIGIT_DETECT;
} else {
dsp_features &= ~DSP_FEATURE_DIGIT_DETECT;
}
if (dsp_features) {
if (!channel->session->dsp) {
if (!(channel->session->dsp = ast_dsp_new())) {
ast_channel_unlock(chan);
return 0;
}
}
ast_dsp_set_features(channel->session->dsp, dsp_features);
} else if (channel->session->dsp) {
ast_dsp_free(channel->session->dsp);
channel->session->dsp = NULL;
}
ast_channel_unlock(chan);
return ast_sip_push_task_synchronous(channel->session->serializer, dtmf_mode_refresh_cb, &rdata);
}
static int refresh_write_cb(void *obj)
{
struct refresh_data *data = obj;

25
channels/pjsip/include/dialplan_functions.h
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@ -47,6 +47,31 @@ int pjsip_acf_channel_read(struct ast_channel *chan, const char *cmd, char *data
*/
int pjsip_acf_media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);
/*!
* \brief PJSIP_DTMF_MODE function read callback
* \param chan The channel the function is called on
* \param cmd The name of the function
* \param data Arguments passed to the function
* \param buf Out buffer that should be populated with the data
* \param len Size of the buffer
*
* \retval 0 on success
* \retval -1 on failure
*/
int pjsip_acf_dtmf_mode_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
/*!
* \brief PJSIP_DTMF_MODE function write callback
* \param chan The channel the function is called on
* \param cmd The name of the function
* \param data Arguments passed to the function
* \param value Value to be set by the function
*
* \retval 0 on success
* \retval -1 on failure
*/
int pjsip_acf_dtmf_mode_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);
/*!
* \brief PJSIP_MEDIA_OFFER function read callback
* \param chan The channel the function is called on

27
include/asterisk/res_pjsip.h
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@ -2859,4 +2859,31 @@ int ast_sip_set_tpselector_from_ep_or_uri(const struct ast_sip_endpoint *endpoin
int ast_sip_dlg_set_transport(const struct ast_sip_endpoint *endpoint, pjsip_dialog *dlg,
pjsip_tpselector *selector);
/*!
* \brief Convert the DTMF mode enum value into a string
* \since 13.18.0
*
* \param dtmf the dtmf mode
* \param buf Buffer to receive dtmf mode string
* \param buf_len Buffer length
*
* \retval 0 Success
* \retval -1 Failure
*
*/
int ast_sip_dtmf_to_str(const enum ast_sip_dtmf_mode dtmf,
char *buf, size_t buf_len);
/*!
* \brief Convert the DTMF mode name into an enum
* \since 13.18.0
*
* \param dtmf_mode dtmf mode as a string
*
* \retval >= 0 The enum value
* \retval -1 Failure
*
*/
int ast_sip_str_to_dtmf(const char *dtmf_mode);
#endif /* _RES_PJSIP_H */

2
include/asterisk/res_pjsip_session.h
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@ -159,6 +159,8 @@ struct ast_sip_session {
unsigned int defer_end:1;
/*! Session end (remote hangup) requested while termination deferred */
unsigned int ended_while_deferred:1;
/*! DTMF mode to use with this session, from endpoint but can change */
enum ast_sip_dtmf_mode dtmf;
};
typedef int (*ast_sip_session_request_creation_cb)(struct ast_sip_session *session, pjsip_tx_data *tdata);

50
res/res_pjsip.c
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@ -4431,6 +4431,56 @@ const char *ast_sip_get_host_ip_string(int af)
return NULL;
}
int ast_sip_dtmf_to_str(const enum ast_sip_dtmf_mode dtmf,
char *buf, size_t buf_len)
{
switch (dtmf) {
case AST_SIP_DTMF_NONE:
ast_copy_string(buf, "none", buf_len);
break;
case AST_SIP_DTMF_RFC_4733:
ast_copy_string(buf, "rfc4733", buf_len);
break;
case AST_SIP_DTMF_INBAND:
ast_copy_string(buf, "inband", buf_len);
break;
case AST_SIP_DTMF_INFO:
ast_copy_string(buf, "info", buf_len);
break;
case AST_SIP_DTMF_AUTO:
ast_copy_string(buf, "auto", buf_len);
break;
case AST_SIP_DTMF_AUTO_INFO:
ast_copy_string(buf, "auto_info", buf_len);
break;
default:
buf[0] = '\0';
return -1;
}
return 0;
}
int ast_sip_str_to_dtmf(const char * dtmf_mode)
{
int result = -1;
if (!strcasecmp(dtmf_mode, "info")) {
result = AST_SIP_DTMF_INFO;
} else if (!strcasecmp(dtmf_mode, "rfc4733")) {
result = AST_SIP_DTMF_RFC_4733;
} else if (!strcasecmp(dtmf_mode, "inband")) {
result = AST_SIP_DTMF_INBAND;
} else if (!strcasecmp(dtmf_mode, "none")) {
result = AST_SIP_DTMF_NONE;
} else if (!strcasecmp(dtmf_mode, "auto")) {
result = AST_SIP_DTMF_AUTO;
} else if (!strcasecmp(dtmf_mode, "auto_info")) {
result = AST_SIP_DTMF_AUTO_INFO;
}
return result;
}
/*!
* \brief Set name and number information on an identity header.
*

40
res/res_pjsip/pjsip_configuration.c
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@ -365,47 +365,29 @@ static int contact_acl_to_str(const void *obj, const intptr_t *args, char **buf)
static int dtmf_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
{
struct ast_sip_endpoint *endpoint = obj;
enum ast_sip_dtmf_mode dtmf = ast_sip_str_to_dtmf(var->value);
if (!strcasecmp(var->value, "rfc4733")) {
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")) {
endpoint->dtmf = AST_SIP_DTMF_AUTO;
} else if (!strcasecmp(var->value, "none")) {
endpoint->dtmf = AST_SIP_DTMF_NONE;
} else {
if (dtmf == -1) {
return -1;
}
endpoint->dtmf = dtmf;
return 0;
}
static int dtmf_to_str(const void *obj, const intptr_t *args, char **buf)
{
const struct ast_sip_endpoint *endpoint = obj;
char dtmf_str[20];
int result = -1;
switch (endpoint->dtmf) {
case AST_SIP_DTMF_RFC_4733 :
*buf = "rfc4733"; break;
case AST_SIP_DTMF_INBAND :
*buf = "inband"; break;
case AST_SIP_DTMF_INFO :
*buf = "info"; break;
case AST_SIP_DTMF_AUTO :
*buf = "auto"; break;
case AST_SIP_DTMF_AUTO_INFO :
*buf = "auto_info";
break;
default:
*buf = "none";
}
result = ast_sip_dtmf_to_str(endpoint->dtmf, dtmf_str, sizeof(dtmf_str));
*buf = ast_strdup(*buf);
if (result == 0) {
*buf = ast_strdup(dtmf_str);
} else {
*buf = ast_strdup("none");
}
return 0;
}

12
res/res_pjsip_sdp_rtp.c
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@ -247,10 +247,10 @@ 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 || session->endpoint->dtmf == AST_SIP_DTMF_AUTO_INFO) {
if (session->dtmf == AST_SIP_DTMF_RFC_4733 || session->dtmf == AST_SIP_DTMF_AUTO || session->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) {
} else if (session->dtmf == AST_SIP_DTMF_INBAND) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
}
@ -333,11 +333,11 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
}
}
}
if (!tel_event && (session->endpoint->dtmf == AST_SIP_DTMF_AUTO)) {
if (!tel_event && (session->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 (session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
if (tel_event) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
} else {
@ -450,7 +450,7 @@ static int set_caps(struct ast_sip_session *session,
ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
}
if ( ((session->endpoint->dtmf == AST_SIP_DTMF_AUTO) || (session->endpoint->dtmf == AST_SIP_DTMF_AUTO_INFO) )
if ( ((session->dtmf == AST_SIP_DTMF_AUTO) || (session->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);
@ -1178,7 +1178,7 @@ 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 || session->endpoint->dtmf == AST_SIP_DTMF_AUTO_INFO) ? AST_RTP_DTMF : 0;
int noncodec = (session->dtmf == AST_SIP_DTMF_RFC_4733 || session->dtmf == AST_SIP_DTMF_AUTO || session->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);

2
res/res_pjsip_session.c
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@ -1481,6 +1481,8 @@ struct ast_sip_session *ast_sip_session_alloc(struct ast_sip_endpoint *endpoint,
session->contact = ao2_bump(contact);
session->inv_session = inv_session;
session->dtmf = endpoint->dtmf;
if (add_supplements(session)) {
/* Release the ref held by session->inv_session */
ao2_ref(session, -1);

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