Fixed #1626: Assertion when remote sends re-INVITE with less media
Included in the fix is sipp testing scenario.
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@4373 74dad513-b988-da41-8d7b-12977e46ad98
diff --git a/pjsip/src/pjsua-lib/pjsua_media.c b/pjsip/src/pjsua-lib/pjsua_media.c
index bc97915..3b9210b 100644
--- a/pjsip/src/pjsua-lib/pjsua_media.c
+++ b/pjsip/src/pjsua-lib/pjsua_media.c
@@ -1457,12 +1457,12 @@
/* Clean up media transports in provisional media that is not used
* by call media.
*/
-void pjsua_media_prov_clean_up(pjsua_call_id call_id)
+static void media_prov_clean_up(pjsua_call_id call_id, int idx)
{
pjsua_call *call = &pjsua_var.calls[call_id];
unsigned i;
- for (i = 0; i < call->med_prov_cnt; ++i) {
+ for (i = idx; i < call->med_prov_cnt; ++i) {
pjsua_call_media *call_med = &call->media_prov[i];
unsigned j;
pj_bool_t used = PJ_FALSE;
@@ -1489,6 +1489,11 @@
}
}
+void pjsua_media_prov_clean_up(pjsua_call_id call_id)
+{
+ media_prov_clean_up(call_id, 0);
+}
+
pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
pjsip_role_e role,
@@ -1902,6 +1907,8 @@
if (rem_sdp && mi >= rem_sdp->media_count) {
/* Remote might have removed some media lines. */
+ media_prov_clean_up(call->index, rem_sdp->media_count);
+ call->med_prov_cnt = rem_sdp->media_count;
break;
}
diff --git a/tests/pjsua/scripts-sipp/uas-reinv-with-less-media.py b/tests/pjsua/scripts-sipp/uas-reinv-with-less-media.py
new file mode 100644
index 0000000..42b2b88
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/uas-reinv-with-less-media.py
@@ -0,0 +1,8 @@
+# $Id$
+#
+import inc_const as const
+
+PJSUA = ["--null-audio --extra-audio --max-calls=1 $SIPP_URI"]
+
+# Send hold after remote holds (double hold)
+PJSUA_EXPECTS = [[0, const.MEDIA_HOLD, "H"]]
diff --git a/tests/pjsua/scripts-sipp/uas-reinv-with-less-media.xml b/tests/pjsua/scripts-sipp/uas-reinv-with-less-media.xml
new file mode 100644
index 0000000..33ae6a2
--- /dev/null
+++ b/tests/pjsua/scripts-sipp/uas-reinv-with-less-media.xml
@@ -0,0 +1,155 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+<!-- -->
+<!-- Sipp default 'uas' scenario. -->
+<!-- -->
+
+<scenario name="Sending OK and re-INVITE with less media (#16xx)">
+ <!-- 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 request="INVITE" crlf="true">
+ <action>
+ <ereg regexp=".*" search_in="hdr" header="From" assign_to="3"/>
+ <ereg regexp="sip:(.*)>" search_in="hdr" header="Contact" assign_to="4,5"/>
+ <assign assign_to="4" variable="5" />
+ </action>
+ </recv>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 3441953879 3441953879 IN IP4 192.168.0.15
+ s=pjmedia
+ c=IN IP4 192.168.0.15
+ t=0 0
+ m=audio 4000 RTP/AVP 0 96
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:96 telephone-event/8000
+ a=sendrecv
+
+ ]]>
+ </send>
+
+ <recv request="ACK" crlf="true">
+ </recv>
+
+ <pause milliseconds="2000"/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:[$5] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-same-branch
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
+ To[$3]
+ Call-ID: [call_id]
+ Cseq: 2 INVITE
+ Max-Forwards: 70
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 3441953879 3441953879 IN IP4 192.168.0.15
+ s=pjmedia
+ c=IN IP4 192.168.0.15
+ t=0 0
+ m=audio 4000 RTP/AVP 0 96
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:96 telephone-event/8000
+ a=sendonly
+
+ ]]>
+ </send>
+
+ <recv response="200" rtd="true">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:[$5] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-same-branch
+ From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
+ To[$3]
+ Call-ID: [call_id]
+ Cseq: 1 ACK
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="INVITE" crlf="true">
+ </recv>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: sip:sipp@[local_ip]:[local_port]
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=- 3441953879 3441953879 IN IP4 192.168.0.15
+ s=pjmedia
+ c=IN IP4 192.168.0.15
+ t=0 0
+ m=audio 4000 RTP/AVP 0 96
+ a=rtpmap:0 PCMU/8000
+ a=rtpmap:96 telephone-event/8000
+ a=inactive
+
+ ]]>
+ </send>
+
+ <recv request="ACK" crlf="true">
+ </recv>
+
+ <!-- Keep the call open for a while in case the 200 is lost to be -->
+ <!-- able to retransmit it if we receive the BYE again. -->
+ <pause milliseconds="4000"/>
+
+
+ <!-- definition of the response time repartition table (unit is ms) -->
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <!-- definition of the call length repartition table (unit is ms) -->
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+