[Asterisk-code-review] chan_pjsip: Transmit REFER waits for the REFER result setting TRANSF... (...asterisk[16])
George Joseph
asteriskteam at digium.com
Tue Jul 23 09:18:32 CDT 2019
George Joseph has submitted this change and it was merged. ( https://gerrit.asterisk.org/c/asterisk/+/11243 )
Change subject: chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS
......................................................................
chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS
Previously, when a Transfer (REFER) was performed, chan_pjsip would set
the TRANSFERSTATUS to SUCCESS when the REFER was queued up. This did not
reflect a successful/unsuccessful transfer the way chan_sip did.
Added a callback module to process the refer subscription information.
Now depends on res_pjsip_pubsub so call transfer progress can be monitored
and reported
ASTERISK-26968 #close
Reported-by: Dan Cropp
Change-Id: If6c27c757c66f71e8b75e3fe49da53ebe62395dc
---
M channels/chan_pjsip.c
A doc/UPGRADE-staging/chan_pjsip_refer_fix.txt
2 files changed, 143 insertions(+), 3 deletions(-)
Approvals:
George Joseph: Looks good to me, approved; Approved for Submit
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index d1f7b6a..8ab1549 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -28,6 +28,7 @@
/*** MODULEINFO
<depend>pjproject</depend>
<depend>res_pjsip</depend>
+ <depend>res_pjsip_pubsub</depend>
<depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
@@ -1894,6 +1895,130 @@
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
}
+/*! \brief REFER Callback module, used to attach session data structure to subscription */
+static pjsip_module refer_callback_module = {
+ .name = { "REFER Callback", 14 },
+ .id = -1,
+};
+
+/*!
+ * \brief Callback function to report status of implicit REFER-NOTIFY subscription.
+ *
+ * This function will be called on any state change in the REFER-NOTIFY subscription.
+ * Its primary purpose is to report SUCCESS/FAILURE of a transfer initiated via
+ * \ref transfer_refer as well as to terminate the subscription, if necessary.
+ */
+static void xfer_client_on_evsub_state(pjsip_evsub *sub, pjsip_event *event)
+{
+ struct ast_sip_session *session;
+ struct ast_channel *chan = NULL;
+ enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
+ int res = 0;
+
+ if (!event) {
+ return;
+ }
+
+ session = pjsip_evsub_get_mod_data(sub, refer_callback_module.id);
+ if (!session) {
+ return;
+ }
+
+ chan = session->channel;
+ if (!chan) {
+ return;
+ }
+
+ if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACCEPTED) {
+ /* Check if subscription is suppressed and terminate and send completion code, if so. */
+ pjsip_rx_data *rdata;
+ pjsip_generic_string_hdr *refer_sub;
+ const pj_str_t REFER_SUB = { "Refer-Sub", 9 };
+
+ ast_debug(3, "Transfer accepted on channel %s\n", ast_channel_name(chan));
+
+ /* Check if response message */
+ if (event->type == PJSIP_EVENT_TSX_STATE && event->body.tsx_state.type == PJSIP_EVENT_RX_MSG) {
+ rdata = event->body.tsx_state.src.rdata;
+
+ /* Find Refer-Sub header */
+ refer_sub = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &REFER_SUB, NULL);
+
+ /* Check if subscription is suppressed. If it is, the far end will not terminate it,
+ * and the subscription will remain active until it times out. Terminating it here
+ * eliminates the unnecessary timeout.
+ */
+ if (refer_sub && !pj_stricmp2(&refer_sub->hvalue, "false")) {
+ /* Since no subscription is desired, assume that call has been transferred successfully. */
+ /* Terminate subscription. */
+ pjsip_evsub_terminate(sub, PJ_TRUE);
+ res = -1;
+ }
+ }
+ } else if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACTIVE ||
+ pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
+ /* Check for NOTIFY complete or error. */
+ pjsip_msg *msg;
+ pjsip_msg_body *body;
+ pjsip_status_line status_line = { .code = PJSIP_SC_NULL };
+ pj_bool_t is_last;
+ pj_status_t status;
+
+ if (event->type == PJSIP_EVENT_TSX_STATE && event->body.tsx_state.type == PJSIP_EVENT_RX_MSG) {
+ pjsip_rx_data *rdata;
+
+ rdata = event->body.tsx_state.src.rdata;
+ msg = rdata->msg_info.msg;
+
+ if (!pjsip_method_cmp(&msg->line.req.method, pjsip_get_notify_method())) {
+ body = msg->body;
+ if (body && !pj_stricmp2(&body->content_type.type, "message")
+ && !pj_stricmp2(&body->content_type.subtype, "sipfrag")) {
+ pjsip_parse_status_line((char *)body->data, body->len, &status_line);
+ }
+ }
+ } else {
+ status_line.code = 500;
+ status_line.reason = *pjsip_get_status_text(500);
+ }
+
+ is_last = (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED);
+ /* If the status code is >= 200, the subscription is finished. */
+ if (status_line.code >= 200 || is_last) {
+ res = -1;
+
+ /* If the subscription has terminated, return AST_TRANSFER_SUCCESS for 2XX.
+ * Any other status code returns AST_TRANSFER_FAILED.
+ * The subscription should not terminate for any code < 200,
+ * but if it does, that constitutes a failure. */
+ if (status_line.code < 200 || status_line.code >= 300) {
+ message = AST_TRANSFER_FAILED;
+ }
+ /* If subscription not terminated and subscription is finished (status code >= 200)
+ * terminate it */
+ if (!is_last) {
+ pjsip_tx_data *tdata;
+
+ status = pjsip_evsub_initiate(sub, pjsip_get_subscribe_method(), 0, &tdata);
+ if (status == PJ_SUCCESS) {
+ pjsip_evsub_send_request(sub, tdata);
+ }
+ }
+ /* Finished. Remove session from subscription */
+ pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
+ ast_debug(3, "Transfer channel %s completed: %d %.*s (%s)\n",
+ ast_channel_name(chan),
+ status_line.code,
+ (int)status_line.reason.slen, status_line.reason.ptr,
+ (message == AST_TRANSFER_SUCCESS) ? "Success" : "Failure");
+ }
+ }
+
+ if (res) {
+ ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
+ }
+}
+
static void transfer_refer(struct ast_sip_session *session, const char *target)
{
pjsip_evsub *sub;
@@ -1902,14 +2027,20 @@
pjsip_tx_data *packet;
const char *ref_by_val;
char local_info[pj_strlen(&session->inv_session->dlg->local.info_str) + 1];
+ struct pjsip_evsub_user xfer_cb;
- if (pjsip_xfer_create_uac(session->inv_session->dlg, NULL, &sub) != PJ_SUCCESS) {
+ pj_bzero(&xfer_cb, sizeof(xfer_cb));
+ xfer_cb.on_evsub_state = &xfer_client_on_evsub_state;
+
+ if (pjsip_xfer_create_uac(session->inv_session->dlg, &xfer_cb, &sub) != PJ_SUCCESS) {
message = AST_TRANSFER_FAILED;
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
return;
}
+ pjsip_evsub_set_mod_data(sub, refer_callback_module.id, session);
+
if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {
message = AST_TRANSFER_FAILED;
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
@@ -1927,7 +2058,6 @@
}
pjsip_xfer_send_request(sub, packet);
- ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
}
static int transfer(void *data)
@@ -3144,6 +3274,8 @@
goto end;
}
+ ast_sip_register_service(&refer_callback_module);
+
ast_sip_session_register_supplement(&chan_pjsip_supplement);
ast_sip_session_register_supplement(&chan_pjsip_supplement_response);
@@ -3180,6 +3312,7 @@
ast_sip_session_unregister_supplement(&chan_pjsip_supplement_response);
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
ast_sip_session_unregister_supplement(&call_pickup_supplement);
+ ast_sip_unregister_service(&refer_callback_module);
ast_custom_function_unregister(&dtmf_mode_function);
ast_custom_function_unregister(&media_offer_function);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
@@ -3205,6 +3338,8 @@
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
ast_sip_session_unregister_supplement(&call_pickup_supplement);
+ ast_sip_unregister_service(&refer_callback_module);
+
ast_custom_function_unregister(&dtmf_mode_function);
ast_custom_function_unregister(&media_offer_function);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
@@ -3223,5 +3358,5 @@
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
- .requires = "res_pjsip,res_pjsip_session",
+ .requires = "res_pjsip,res_pjsip_session,res_pjsip_pubsub",
);
diff --git a/doc/UPGRADE-staging/chan_pjsip_refer_fix.txt b/doc/UPGRADE-staging/chan_pjsip_refer_fix.txt
new file mode 100644
index 0000000..301930f
--- /dev/null
+++ b/doc/UPGRADE-staging/chan_pjsip_refer_fix.txt
@@ -0,0 +1,5 @@
+Subject: chan_pjsip
+Subject: Core
+
+res_pjsip_pubsub is now required so call transfer progress can be monitored
+and reported in the channel variable TRANSFERSTATUS.
--
To view, visit https://gerrit.asterisk.org/c/asterisk/+/11243
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Gerrit-Project: asterisk
Gerrit-Branch: 16
Gerrit-Change-Id: If6c27c757c66f71e8b75e3fe49da53ebe62395dc
Gerrit-Change-Number: 11243
Gerrit-PatchSet: 6
Gerrit-Owner: Dan Cropp <dan at amtelco.com>
Gerrit-Reviewer: Dan Cropp <dan at amtelco.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Matthew Fredrickson <creslin at digium.com>
Gerrit-MessageType: merged
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