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channels/chan_pjsip: Add PJSIP_SEND_SESSION_REFRESH

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This patch adds a new PJSIP specific dialplan function,
PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media
session will be refreshed via either an UPDATE or re-INVITE request.
When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function,
the formats in use on a PJSIP channel can be re-negotiated and changed
dynamically after call setup.

ASTERISK-26277 #close

Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b
(cherry picked from commit eec60dd773)
changes/52/3452/1
Matt Jordan 9 years ago
parent d78fe8fed0
commit c315460abb
4 changed files with 170 additions and 21 deletions
Show Stats Download Patch File Download Diff File

51
CHANGES
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@ -12,6 +12,37 @@
--- Functionality changes from Asterisk 14 to Asterisk 15 --------------------
------------------------------------------------------------------------------
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ----------
------------------------------------------------------------------------------
app_voicemail
------------------
* Added "tps_queue_high" and "tps_queue_low" options.
The options can modify the taskprocessor alert levels for this module.
Additional information can be found in the sample configuration file at
config/samples/voicemail.conf.sample.
res_pjsip_mwi
------------------
* Added "mwi_tps_queue_high" and "mwi_tps_queue_low" global configuration
options to tune taskprocessor alert levels.
* Added "mwi_disable_initial_unsolicited" global configuration option
to disable sending unsolicited MWI to all endpoints on startup.
Additional information can be found in the sample configuration file at
config/samples/pjsip.conf.sample.
chan_pjsip
------------------
* A new dialplan function, PJSIP_SEND_SESSION_REFRESH, has been added. When
invoked, a re-INVITE or UPDATE request will be sent immediately to the
endpoint underlying the channel. When used in combination with the existing
dialplan function PJSIP_MEDIA_OFFER, this allows the formats on a PJSIP
channel to be re-negotiated and updated after session set up.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13 to Asterisk 14 --------------------
------------------------------------------------------------------------------
@ -378,26 +409,6 @@ cdr_csv
post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.11.0 to Asterisk 13.12.0 ----------
------------------------------------------------------------------------------
app_voicemail
------------------
* Added "tps_queue_high" and "tps_queue_low" options.
The options can modify the taskprocessor alert levels for this module.
Additional information can be found in the sample configuration file at
config/samples/voicemail.conf.sample.
res_pjsip_mwi
------------------
* Added "mwi_tps_queue_high" and "mwi_tps_queue_low" global configuration
options to tune taskprocessor alert levels.
* Added "mwi_disable_initial_unsolicited" global configuration option
to disable sending unsolicited MWI to all endpoints on startup.
Additional information can be found in the sample configuration file at
config/samples/pjsip.conf.sample.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.10.0 to Asterisk 13.11.0 ----------
------------------------------------------------------------------------------

12
channels/chan_pjsip.c
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@ -2433,6 +2433,11 @@ static struct ast_custom_function media_offer_function = {
.write = pjsip_acf_media_offer_write
};
static struct ast_custom_function session_refresh_function = {
.name = "PJSIP_SEND_SESSION_REFRESH",
.write = pjsip_acf_session_refresh_write,
};
/*!
* \brief Load the module
*
@ -2472,6 +2477,11 @@ static int load_module(void)
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;
}
if (ast_sip_session_register_supplement(&chan_pjsip_supplement)) {
ast_log(LOG_ERROR, "Unable to register PJSIP supplement\n");
goto end;
@ -2528,6 +2538,7 @@ end:
pjsip_uids_onhold = NULL;
ast_custom_function_unregister(&media_offer_function);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
ast_custom_function_unregister(&session_refresh_function);
ast_channel_unregister(&chan_pjsip_tech);
ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
@ -2549,6 +2560,7 @@ static int unload_module(void)
ast_custom_function_unregister(&media_offer_function);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
ast_custom_function_unregister(&session_refresh_function);
ast_channel_unregister(&chan_pjsip_tech);
ao2_ref(chan_pjsip_tech.capabilities, -1);

116
channels/pjsip/dialplan_functions.c
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@ -59,8 +59,55 @@
</parameter>
</syntax>
<description>
<para>Returns the codecs offered based upon the media choice</para>
<para>When read, returns the codecs offered based upon the media choice.</para>
<para>When written, sets the codecs to offer when an outbound dial attempt is made,
or when a session refresh is sent using <replaceable>PJSIP_SEND_SESSION_REFRESH</replaceable>.
</para>
</description>
<see-also>
<ref type="function">PJSIP_SEND_SESSION_REFRESH</ref>
</see-also>
</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
</synopsis>
<syntax>
<parameter name="update_type" required="false">
<para>The type of update to send. Default is <literal>invite</literal>.</para>
<enumlist>
<enum name="invite">
<para>Send the session refresh as a re-INVITE.</para>
</enum>
<enum name="update">
<para>Send the session refresh as an UPDATE.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>This function will cause the PJSIP stack to immediately refresh
the media session for the channel. This will be done using either a
re-INVITE (default) or an UPDATE request.
</para>
<para>This is most useful when combined with the <replaceable>PJSIP_MEDIA_OFFER</replaceable>
dialplan function, as it allows the formats in use on a channel to be
re-negotiated after call setup.</para>
<warning>
<para>The formats the endpoint supports are <emphasis>not</emphasis>
checked or enforced by this function. Using this function to offer
formats not supported by the endpoint <emphasis>may</emphasis> result
in a loss of media.</para>
</warning>
<example title="Re-negotiate format to g722">
; Within some existing extension on an answered channel
same => n,Set(PJSIP_MEDIA_OFFER(audio)=!all,g722)
same => n,Set(PJSIP_SEND_SESSION_REFRESH()=invite)
</example>
</description>
<see-also>
<ref type="function">PJSIP_MEDIA_OFFER</ref>
</see-also>
</function>
<info name="PJSIPCHANNEL" language="en_US" tech="PJSIP">
<enumlist>
@ -961,3 +1008,70 @@ 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);
}
struct refresh_data {
struct ast_sip_session *session;
enum ast_sip_session_refresh_method method;
};
static int sip_session_response_cb(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
struct ast_format *fmt;
if (!session->channel) {
/* Egads! */
return 0;
}
fmt = ast_format_cap_get_best_by_type(ast_channel_nativeformats(session->channel), AST_MEDIA_TYPE_AUDIO);
if (!fmt) {
/* No format? That's weird. */
return 0;
}
ast_channel_set_writeformat(session->channel, fmt);
ast_channel_set_rawwriteformat(session->channel, fmt);
ast_channel_set_readformat(session->channel, fmt);
ast_channel_set_rawreadformat(session->channel, fmt);
ao2_ref(fmt, -1);
return 0;
}
static int refresh_write_cb(void *obj)
{
struct refresh_data *data = obj;
ast_sip_session_refresh(data->session, NULL, NULL,
sip_session_response_cb, data->method, 1);
return 0;
}
int pjsip_acf_session_refresh_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct ast_sip_channel_pvt *channel;
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;
}
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
return -1;
}
channel = ast_channel_tech_pvt(chan);
rdata.session = channel->session;
if (!strcmp(value, "invite")) {
rdata.method = AST_SIP_SESSION_REFRESH_METHOD_INVITE;
} else if (!strcmp(value, "update")) {
rdata.method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
}
return ast_sip_push_task_synchronous(channel->session->serializer, refresh_write_cb, &rdata);
}

12
channels/pjsip/include/dialplan_functions.h
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@ -60,6 +60,18 @@ int pjsip_acf_media_offer_write(struct ast_channel *chan, const char *cmd, char
*/
int pjsip_acf_media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
/*!
* \brief PJSIP_SEND_SESSION_REFRESH 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_session_refresh_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);
/*!
* \brief PJSIP_DIAL_CONTACTS function read callback
* \param chan The channel the function is called on

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