Hello,
Since 4b36447313032c1aaa0a68a91066b44e86c5e466 there has been a dialog memory leak when an INVITE is received without an SDP. I have attached a patch (against master) that fixes the leak.
For testing I used a simple pjsua UAS program (attached -- repro.c) and two different SIPp[1] scenarios (one with SDP in INVITE and one with SDP later in the ACK -- both attached). After the final transaction is terminated 30 seconds after the call ends (typically when the dialog would get destroyed) you can send SIGUSR2 to the pjsua UAS program to display the dialog sets still in memory. Before this patch, the sip-ack scenario should show a single dialog set. With the sipp-invite (and after this patch + sipp-ack) scenario, there should be no dialog sets shown.
The SIPp scenarios can be used via: sipp -sf <scenario file> -m 1 -r 1 -l 1 <ip where pjsua UAS program is running>:5070
From b09bf02c6ed1c74d3006cd914aaed0ce83ce5a20 Mon Sep 17 00:00:00 2001
From: mscdex <mscdex@xxxxxxxxxx>
Date: Fri, 5 Apr 2019 13:09:35 -0400
Subject: [PATCH] Fixed dialog memory leak when no SDP in incoming INVITE
---
pjsip/src/pjsua-lib/pjsua_call.c | 12 ++++++++++++
1 file changed, 12 insertions(+)
diff --git a/pjsip/src/pjsua-lib/pjsua_call.c b/pjsip/src/pjsua-lib/pjsua_call.c
index 74041065..e0bb8f76 100644
--- a/pjsip/src/pjsua-lib/pjsua_call.c
+++ b/pjsip/src/pjsua-lib/pjsua_call.c
@@ -2407,11 +2407,20 @@ on_answer_call_med_tp_complete(pjsua_call_id call_id,
{
pjsua_call *call = &pjsua_var.calls[call_id];
pjmedia_sdp_session *sdp;
+ pjsip_dialog *dlg = call->async_call.dlg;
int sip_err_code = (info? info->sip_err_code: 0);
pj_status_t status = (info? info->status: PJ_SUCCESS);
PJSUA_LOCK();
+ /* Increment the dialog's lock to prevent it to be destroyed prematurely,
+ * such as in case of transport error.
+ */
+ pjsip_dlg_inc_lock(dlg);
+
+ /* Decrement dialog session. */
+ pjsip_dlg_dec_session(dlg, &pjsua_var.mod);
+
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error initializing media channel", status);
goto on_return;
@@ -2421,6 +2430,7 @@ on_answer_call_med_tp_complete(pjsua_call_id call_id,
if (call->async_call.med_ch_deinit) {
pjsua_media_channel_deinit(call->index);
call->med_ch_cb = NULL;
+ pjsip_dlg_dec_lock(dlg);
PJSUA_UNLOCK();
return PJ_SUCCESS;
}
@@ -2470,6 +2480,8 @@ on_return:
process_pending_call_answer(call);
}
+ pjsip_dlg_dec_lock(dlg);
+
PJSUA_UNLOCK();
return status;
}
--
2.17.1
// gcc repro.c -o repro -g -L/usr/local/lib -lpjsua -lpjsip-ua -lpjsip-simple -lpjsip -lpjmedia-codec -lpjmedia -lpjmedia-videodev -lpjmedia-audiodev -lpjmedia -lpjnath -lpjlib-util -lsrtp -lresample -lilbccodec -lpj -lm -lrt -lpthread -lcrypto -lssl
#include <signal.h>
#include <pjsua-lib/pjsua.h>
#define THIS_FILE "APP"
static void on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id,
pjsip_rx_data *rdata) {
pjsua_call_info ci;
PJ_UNUSED_ARG(acc_id);
PJ_UNUSED_ARG(rdata);
pjsua_call_get_info(call_id, &ci);
PJ_LOG(3,(THIS_FILE, "Incoming call from %.*s!!",
(int)ci.remote_info.slen,
ci.remote_info.ptr));
/* Automatically answer incoming calls with 200/OK */
pjsua_call_answer(call_id, 200, NULL, NULL);
}
static void on_call_state(pjsua_call_id call_id, pjsip_event *e) {
pjsua_call_info ci;
PJ_UNUSED_ARG(e);
pjsua_call_get_info(call_id, &ci);
PJ_LOG(3,(THIS_FILE, "Call %d state=%.*s", call_id,
(int)ci.state_text.slen,
ci.state_text.ptr));
}
static void error_exit(const char *title, pj_status_t status) {
pjsua_perror(THIS_FILE, title, status);
pjsua_destroy();
exit(1);
}
void sighandler(int signum, siginfo_t* info, void* ptr) {
pjsua_dump(PJ_TRUE);
}
int main(int argc, char *argv[]) {
setbuf(stdout, NULL);
setbuf(stderr, NULL);
pjsua_acc_id acc_id;
pj_status_t status;
/* Create pjsua first! */
status = pjsua_create();
if (status != PJ_SUCCESS)
error_exit("Error in pjsua_create()", status);
/* Init pjsua */
{
pjsua_config cfg;
pjsua_logging_config log_cfg;
pjsua_config_default(&cfg);
cfg.cb.on_incoming_call = &on_incoming_call;
cfg.cb.on_call_state = &on_call_state;
pjsua_logging_config_default(&log_cfg);
log_cfg.console_level = log_cfg.level = 5;
status = pjsua_init(&cfg, &log_cfg, NULL);
if (status != PJ_SUCCESS)
error_exit("Error in pjsua_init()", status);
}
/* Add UDP transport. */
{
pjsua_transport_config cfg;
pjsua_transport_config_default(&cfg);
cfg.port = 5070;
status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);
if (status != PJ_SUCCESS)
error_exit("Error creating transport", status);
}
/* Initialization is done, now start pjsua */
status = pjsua_start();
if (status != PJ_SUCCESS)
error_exit("Error starting pjsua", status);
/* Register to SIP server by creating SIP account. */
{
pjsua_acc_config cfg;
pjsua_acc_config_default(&cfg);
cfg.id = pj_str("sip:1234@127.0.0.1");
status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
if (status != PJ_SUCCESS)
error_exit("Error adding account", status);
}
struct sigaction act;
act.sa_sigaction = sighandler;
act.sa_flags = SA_SIGINFO;
sigaction(SIGUSR2, &act, NULL);
while (1)
pj_thread_sleep(100);
return 0;
}
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
]]>
</send>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<!--<pause milliseconds="3000"/>-->
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500" crlf="true">
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
</scenario>
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
]]>
</send>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<!--<pause milliseconds="3000"/>-->
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500" crlf="true">
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
</scenario>
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