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Expose the chan_pjsip implementation pvt and session in a defined manner.

Browse Source 
This allows modules outside of chan_pjsip itself to get the session given
only an Asterisk channel.

Review: https://reviewboard.asterisk.org/r/2674/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2013-07-23 12:27:03 +00:00
parent b4c2eecca6
commit 16885ffda5
4 changed files with 129 additions and 82 deletions
Download Patch File Download Diff File Expand all files Collapse all files

164
channels/chan_gulp.c
View File

@@ -114,7 +114,6 @@ enum sip_session_media_position {
};
struct gulp_pvt {
struct ast_sip_session *session;
struct ast_sip_session_media *media[SIP_MEDIA_SIZE];
};
@@ -123,9 +122,6 @@ static void gulp_pvt_dtor(void *obj)
struct gulp_pvt *pvt = obj;
int i;
ao2_cleanup(pvt->session);
pvt->session = NULL;
for (i = 0; i < SIP_MEDIA_SIZE; ++i) {
ao2_cleanup(pvt->media[i]);
pvt->media[i] = NULL;
@@ -336,12 +332,12 @@ static int media_offer_write_av(void *obj)
static int media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
if (!strcmp(data, "audio")) {
return media_offer_read_av(pvt->session, buf, len, AST_FORMAT_TYPE_AUDIO);
return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_AUDIO);
} else if (!strcmp(data, "video")) {
return media_offer_read_av(pvt->session, buf, len, AST_FORMAT_TYPE_VIDEO);
return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_VIDEO);
}
return 0;
@@ -349,10 +345,10 @@ static int media_offer_read(struct ast_channel *chan, const char *cmd, char *dat
static int media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct media_offer_data mdata = {
.session = pvt->session,
.session = channel->session,
.value = value
};
@@ -362,7 +358,7 @@ static int media_offer_write(struct ast_channel *chan, const char *cmd, char *da
mdata.media_type = AST_FORMAT_TYPE_VIDEO;
}
return ast_sip_push_task_synchronous(pvt->session->serializer, media_offer_write_av, &mdata);
return ast_sip_push_task_synchronous(channel->session->serializer, media_offer_write_av, &mdata);
}
static struct ast_custom_function media_offer_function = {
@@ -374,14 +370,15 @@ static struct ast_custom_function media_offer_function = {
/*! \brief Function called by RTP engine to get local audio RTP peer */
static enum ast_rtp_glue_result gulp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct gulp_pvt *pvt = channel->pvt;
struct ast_sip_endpoint *endpoint;
if (!pvt || !pvt->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
return AST_RTP_GLUE_RESULT_FORBID;
}
endpoint = pvt->session->endpoint;
endpoint = channel->session->endpoint;
*instance = pvt->media[SIP_MEDIA_AUDIO]->rtp;
ao2_ref(*instance, +1);
@@ -397,9 +394,10 @@ static enum ast_rtp_glue_result gulp_get_rtp_peer(struct ast_channel *chan, stru
/*! \brief Function called by RTP engine to get local video RTP peer */
static enum ast_rtp_glue_result gulp_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct gulp_pvt *pvt = channel->pvt;
if (!pvt || !pvt->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
return AST_RTP_GLUE_RESULT_FORBID;
}
@@ -412,9 +410,9 @@ static enum ast_rtp_glue_result gulp_get_vrtp_peer(struct ast_channel *chan, str
/*! \brief Function called by RTP engine to get peer capabilities */
static void gulp_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
ast_format_cap_copy(result, pvt->session->endpoint->codecs);
ast_format_cap_copy(result, channel->session->endpoint->codecs);
}
static int send_direct_media_request(void *data)
@@ -486,8 +484,9 @@ static int gulp_set_rtp_peer(struct ast_channel *chan,
const struct ast_format_cap *cap,
int nat_active)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
struct ast_sip_session *session = pvt->session;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct gulp_pvt *pvt = channel->pvt;
struct ast_sip_session *session = channel->session;
int changed = 0;
struct ast_channel *bridge_peer;
@@ -544,7 +543,8 @@ static struct ast_channel *gulp_new(struct ast_sip_session *session, int state,
{
struct ast_channel *chan;
struct ast_format fmt;
struct gulp_pvt *pvt;
RAII_VAR(struct gulp_pvt *, pvt, NULL, ao2_cleanup);
struct ast_sip_channel_pvt *channel;
if (!(pvt = ao2_alloc(sizeof(*pvt), gulp_pvt_dtor))) {
return NULL;
@@ -552,20 +552,22 @@ static struct ast_channel *gulp_new(struct ast_sip_session *session, int state,
if (!(chan = ast_channel_alloc(1, state, S_OR(session->id.number.str, ""), S_OR(session->id.name.str, ""), "", "", "", linkedid, 0, "Gulp/%s-%08x", ast_sorcery_object_get_id(session->endpoint),
ast_atomic_fetchadd_int((int *)&chan_idx, +1)))) {
ao2_cleanup(pvt);
return NULL;
}
ast_channel_tech_set(chan, &gulp_tech);
ao2_ref(session, +1);
pvt->session = session;
if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
ast_hangup(chan);
return NULL;
}
/* If res_sip_session is ever updated to create/destroy ast_sip_session_media
* during a call such as if multiple same-type stream support is introduced,
* these will need to be recaptured as well */
pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio", OBJ_KEY);
pvt->media[SIP_MEDIA_VIDEO] = ao2_find(session->media, "video", OBJ_KEY);
ast_channel_tech_pvt_set(chan, pvt);
ast_channel_tech_pvt_set(chan, channel);
if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(chan));
}
@@ -573,7 +575,6 @@ static struct ast_channel *gulp_new(struct ast_sip_session *session, int state,
ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(chan));
}
if (ast_format_cap_is_empty(session->req_caps) || !ast_format_cap_has_joint(session->req_caps, session->endpoint->codecs)) {
ast_format_cap_copy(ast_channel_nativeformats(chan), session->endpoint->codecs);
} else {
@@ -637,8 +638,7 @@ static int answer(void *data)
/*! \brief Function called by core when we should answer a Gulp session */
static int gulp_answer(struct ast_channel *ast)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
struct ast_sip_session *session = pvt->session;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
if (ast_channel_state(ast) == AST_STATE_UP) {
return 0;
@@ -646,10 +646,10 @@ static int gulp_answer(struct ast_channel *ast)
ast_setstate(ast, AST_STATE_UP);
ao2_ref(session, +1);
if (ast_sip_push_task(session->serializer, answer, session)) {
ao2_ref(channel->session, +1);
if (ast_sip_push_task(channel->session->serializer, answer, channel->session)) {
ast_log(LOG_WARNING, "Unable to push answer task to the threadpool. Cannot answer call\n");
ao2_cleanup(session);
ao2_cleanup(channel->session);
return -1;
}
@@ -659,8 +659,8 @@ static int gulp_answer(struct ast_channel *ast)
/*! \brief Function called by core to read any waiting frames */
static struct ast_frame *gulp_read(struct ast_channel *ast)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
struct ast_sip_session *session = pvt->session;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
struct gulp_pvt *pvt = channel->pvt;
struct ast_frame *f;
struct ast_sip_session_media *media = NULL;
int rtcp = 0;
@@ -702,8 +702,8 @@ static struct ast_frame *gulp_read(struct ast_channel *ast)
ast_set_write_format(ast, ast_channel_writeformat(ast));
}
if (session->dsp) {
f = ast_dsp_process(ast, session->dsp, f);
if (channel->session->dsp) {
f = ast_dsp_process(ast, channel->session->dsp, f);
if (f && (f->frametype == AST_FRAME_DTMF)) {
ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
@@ -717,7 +717,8 @@ static struct ast_frame *gulp_read(struct ast_channel *ast)
/*! \brief Function called by core to write frames */
static int gulp_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
struct gulp_pvt *pvt = channel->pvt;
struct ast_sip_session_media *media;
int res = 0;
@@ -764,9 +765,10 @@ struct fixup_data {
static int fixup(void *data)
{
struct fixup_data *fix_data = data;
struct gulp_pvt *pvt = ast_channel_tech_pvt(fix_data->chan);
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(fix_data->chan);
struct gulp_pvt *pvt = channel->pvt;
fix_data->session->channel = fix_data->chan;
channel->session->channel = fix_data->chan;
if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(fix_data->chan));
}
@@ -780,18 +782,17 @@ static int fixup(void *data)
/*! \brief Function called by core to change the underlying owner channel */
static int gulp_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(newchan);
struct ast_sip_session *session = pvt->session;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);
struct fixup_data fix_data;
fix_data.session = session;
fix_data.session = channel->session;
fix_data.chan = newchan;
if (session->channel != oldchan) {
if (channel->session->channel != oldchan) {
return -1;
}
if (ast_sip_push_task_synchronous(session->serializer, fixup, &fix_data)) {
if (ast_sip_push_task_synchronous(channel->session->serializer, fixup, &fix_data)) {
ast_log(LOG_WARNING, "Unable to perform channel fixup\n");
return -1;
}
@@ -990,8 +991,8 @@ static int update_connected_line_information(void *data)
/*! \brief Function called by core to ask the channel to indicate some sort of condition */
static int gulp_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
struct ast_sip_session *session = pvt->session;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
struct gulp_pvt *pvt = channel->pvt;
struct ast_sip_session_media *media;
int response_code = 0;
int res = 0;
@@ -999,7 +1000,7 @@ static int gulp_indicate(struct ast_channel *ast, int condition, const void *dat
switch (condition) {
case AST_CONTROL_RINGING:
if (ast_channel_state(ast) == AST_STATE_RING) {
if (session->endpoint->inband_progress) {
if (channel->session->endpoint->inband_progress) {
response_code = 183;
res = -1;
} else {
@@ -1008,7 +1009,7 @@ static int gulp_indicate(struct ast_channel *ast, int condition, const void *dat
} else {
res = -1;
}
ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "Gulp/%s", ast_sorcery_object_get_id(session->endpoint));
ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "Gulp/%s", ast_sorcery_object_get_id(channel->session->endpoint));
break;
case AST_CONTROL_BUSY:
if (ast_channel_state(ast) != AST_STATE_UP) {
@@ -1048,19 +1049,19 @@ static int gulp_indicate(struct ast_channel *ast, int condition, const void *dat
case AST_CONTROL_VIDUPDATE:
media = pvt->media[SIP_MEDIA_VIDEO];
if (media && media->rtp) {
ao2_ref(session, +1);
ao2_ref(channel->session, +1);
if (ast_sip_push_task(session->serializer, transmit_info_with_vidupdate, session)) {
ao2_cleanup(session);
if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
ao2_cleanup(channel->session);
}
} else {
res = -1;
}
break;
case AST_CONTROL_CONNECTED_LINE:
ao2_ref(session, +1);
if (ast_sip_push_task(session->serializer, update_connected_line_information, session)) {
ao2_cleanup(session);
ao2_ref(channel->session, +1);
if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) {
ao2_cleanup(channel->session);
}
break;
case AST_CONTROL_UPDATE_RTP_PEER:
@@ -1095,10 +1096,10 @@ static int gulp_indicate(struct ast_channel *ast, int condition, const void *dat
}
if (response_code) {
struct indicate_data *ind_data = indicate_data_alloc(session, condition, response_code, data, datalen);
if (!ind_data || ast_sip_push_task(session->serializer, indicate, ind_data)) {
struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
if (!ind_data || ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
ast_log(LOG_NOTICE, "Cannot send response code %d to endpoint %s. Could not queue task properly\n",
response_code, ast_sorcery_object_get_id(session->endpoint));
response_code, ast_sorcery_object_get_id(channel->session->endpoint));
ao2_cleanup(ind_data);
res = -1;
}
@@ -1214,15 +1215,14 @@ static int transfer(void *data)
/*! \brief Function called by core for Asterisk initiated transfer */
static int gulp_transfer(struct ast_channel *chan, const char *target)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
struct ast_sip_session *session = pvt->session;
struct transfer_data *trnf_data = transfer_data_alloc(session, target);
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
if (!trnf_data) {
return -1;
}
if (ast_sip_push_task(session->serializer, transfer, trnf_data)) {
if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
ast_log(LOG_WARNING, "Error requesting transfer\n");
ao2_cleanup(trnf_data);
return -1;
@@ -1234,12 +1234,12 @@ static int gulp_transfer(struct ast_channel *chan, const char *target)
/*! \brief Function called by core to start a DTMF digit */
static int gulp_digit_begin(struct ast_channel *chan, char digit)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
struct ast_sip_session *session = pvt->session;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct gulp_pvt *pvt = channel->pvt;
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
int res = 0;
switch (session->endpoint->dtmf) {
switch (channel->session->endpoint->dtmf) {
case AST_SIP_DTMF_RFC_4733:
if (!media || !media->rtp) {
return -1;
@@ -1322,21 +1322,21 @@ static int transmit_info_dtmf(void *data)
/*! \brief Function called by core to stop a DTMF digit */
static int gulp_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
struct ast_sip_session *session = pvt->session;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
struct gulp_pvt *pvt = channel->pvt;
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
int res = 0;
switch (session->endpoint->dtmf) {
switch (channel->session->endpoint->dtmf) {
case AST_SIP_DTMF_INFO:
{
struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(session, digit, duration);
struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
if (!dtmf_data) {
return -1;
}
if (ast_sip_push_task(session->serializer, transmit_info_dtmf, dtmf_data)) {
if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
ao2_cleanup(dtmf_data);
return -1;
@@ -1378,13 +1378,12 @@ static int call(void *data)
/*! \brief Function called by core to actually start calling a remote party */
static int gulp_call(struct ast_channel *ast, const char *dest, int timeout)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
struct ast_sip_session *session = pvt->session;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
ao2_ref(session, +1);
if (ast_sip_push_task(session->serializer, call, session)) {
ao2_ref(channel->session, +1);
if (ast_sip_push_task(channel->session->serializer, call, channel->session)) {
ast_log(LOG_WARNING, "Error attempting to place outbound call to call '%s'\n", dest);
ao2_cleanup(session);
ao2_cleanup(channel->session);
return -1;
}
@@ -1484,8 +1483,9 @@ static int hangup(void *data)
pjsip_tx_data *packet = NULL;
struct hangup_data *h_data = data;
struct ast_channel *ast = h_data->chan;
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
struct ast_sip_session *session = pvt->session;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
struct gulp_pvt *pvt = channel->pvt;
struct ast_sip_session *session = channel->session;
int cause = h_data->cause;
if (!session->defer_terminate &&
@@ -1507,16 +1507,16 @@ static int hangup(void *data)
/*! \brief Function called by core to hang up a Gulp session */
static int gulp_hangup(struct ast_channel *ast)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
struct ast_sip_session *session = pvt->session;
int cause = hangup_cause2sip(ast_channel_hangupcause(session->channel));
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
struct gulp_pvt *pvt = channel->pvt;
int cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
struct hangup_data *h_data = hangup_data_alloc(cause, ast);
if (!h_data) {
goto failure;
}
if (ast_sip_push_task(session->serializer, hangup, h_data)) {
if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
goto failure;
}
@@ -1527,7 +1527,7 @@ failure:
/* Go ahead and do our cleanup of the session and channel even if we're not going
* to be able to send our SIP request/response
*/
clear_session_and_channel(session, ast, pvt);
clear_session_and_channel(channel->session, ast, pvt);
ao2_cleanup(pvt);
ao2_cleanup(h_data);
@@ -1665,10 +1665,10 @@ static int sendtext(void *obj)
/*! \brief Function called by core to send text on Gulp session */
static int gulp_sendtext(struct ast_channel *ast, const char *text)
{
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
struct sendtext_data *data = sendtext_data_create(pvt->session, text);
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
struct sendtext_data *data = sendtext_data_create(channel->session, text);
if (!data || ast_sip_push_task(pvt->session->serializer, sendtext, data)) {
if (!data || ast_sip_push_task(channel->session->serializer, sendtext, data)) {
ao2_ref(data, -1);
return -1;
}

21
include/asterisk/res_sip_session.h
View File

@@ -272,6 +272,27 @@ struct ast_sip_session_sdp_handler {
AST_LIST_ENTRY(ast_sip_session_sdp_handler) next;
};
/*!
* \brief A structure which contains a channel implementation and session
*/
struct ast_sip_channel_pvt {
/*! \brief Pointer to channel specific implementation information, must be ao2 object */
void *pvt;
/*! \brief Pointer to session */
struct ast_sip_session *session;
};
/*!
* \brief Allocate a new SIP channel pvt structure
*
* \param pvt Pointer to channel specific implementation
* \param session Pointer to SIP session
*
* \retval non-NULL success
* \retval NULL failure
*/
struct ast_sip_channel_pvt *ast_sip_channel_pvt_alloc(void *pvt, struct ast_sip_session *session);
/*!
* \brief Allocate a new SIP session
*

25
res/res_sip_session.c
View File

@@ -901,6 +901,31 @@ static int add_session_media(void *obj, void *arg, int flags)
return 0;
}
/*! \brief Destructor for SIP channel */
static void sip_channel_destroy(void *obj)
{
struct ast_sip_channel_pvt *channel = obj;
ao2_cleanup(channel->pvt);
ao2_cleanup(channel->session);
}
struct ast_sip_channel_pvt *ast_sip_channel_pvt_alloc(void *pvt, struct ast_sip_session *session)
{
struct ast_sip_channel_pvt *channel = ao2_alloc(sizeof(*channel), sip_channel_destroy);
if (!channel) {
return NULL;
}
ao2_ref(pvt, +1);
channel->pvt = pvt;
ao2_ref(session, +1);
channel->session = session;
return channel;
}
struct ast_sip_session *ast_sip_session_alloc(struct ast_sip_endpoint *endpoint, pjsip_inv_session *inv_session)
{
RAII_VAR(struct ast_sip_session *, session, ao2_alloc(sizeof(*session), session_destructor), ao2_cleanup);

1
res/res_sip_session.exports.in
View File

@@ -16,6 +16,7 @@
LINKER_SYMBOL_PREFIXast_sip_session_create_invite;
LINKER_SYMBOL_PREFIXast_sip_session_create_outgoing;
LINKER_SYMBOL_PREFIXast_sip_dialog_get_session;
LINKER_SYMBOL_PREFIXast_sip_channel_pvt_alloc;
local:
*;
};