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Re #1864: fixed assert and crash when remote remove and add back media lines in SDP: 

- reverted back r5124 (of this ticket) and r4373 (of #1626)
 - applied another fix for #1626 as in described in the ticket desc
 - also fixed a related add/remove media issue:
    - we send an offer with 2 audio and 1 video in SDP
    - remote answers with only 1 audio in SDP
    - when we send a re-offer/re-INVITE, we have 5 media lines in our SDP (3 audio + 2 video, note 1 audio and 1 video are disabled)
    - call settings are actually maintained (2 active audio + 1 active video), just better to avoid the additional disabled media.



git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@5135 74dad513-b988-da41-8d7b-12977e46ad98
This commit is contained in:
Nanang Izzuddin 2015-07-14 08:38:29 +00:00
parent 64d67d7c93
commit bcf74d9be6
5 changed files with 567 additions and 33 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

133
pjsip/src/pjsua-lib/pjsua_media.c
View File

@ -1139,6 +1139,72 @@ static void sort_media(const pjmedia_sdp_session *sdp,
}
}
/* Go through the list of media in the call, find acceptable media, and
* sort them based on the "quality" of the media, and store the indexes
* in the specified array. Media with the best quality will be listed
* first in the array.
*/
static void sort_media2(const pjsua_call_media *call_med,
unsigned call_med_cnt,
pjmedia_type type,
pj_uint8_t midx[],
unsigned *p_count,
unsigned *p_total_count)
{
unsigned i;
unsigned count = 0;
int score[PJSUA_MAX_CALL_MEDIA];
pj_assert(*p_count >= PJSUA_MAX_CALL_MEDIA);
pj_assert(*p_total_count >= PJSUA_MAX_CALL_MEDIA);
*p_count = 0;
*p_total_count = 0;
for (i=0; i<PJSUA_MAX_CALL_MEDIA; ++i)
score[i] = 1;
/* Score each media */
for (i=0; i<call_med_cnt && count<PJSUA_MAX_CALL_MEDIA; ++i) {
/* Skip different media */
if (call_med[i].type != type) {
score[count++] = -22000;
continue;
}
/* Is it active? */
if (!call_med[i].tp) {
score[i] -= 10;
}
++count;
}
/* Created sorted list based on quality */
for (i=0; i<count; ++i) {
unsigned j;
int best = 0;
for (j=1; j<count; ++j) {
if (score[j] > score[best])
best = j;
}
/* Don't put media with negative score, that media is unacceptable
* for us.
*/
midx[i] = (pj_uint8_t)best;
if (score[best] >= 0)
(*p_count)++;
if (score[best] > -22000)
(*p_total_count)++;
score[best] = -22000;
}
}
/* Callback to receive media events */
pj_status_t call_media_on_event(pjmedia_event *event,
void *user_data)
@ -1528,23 +1594,21 @@ on_return:
}
/* If idx == 0, clean up media transports in provisional media that
* is not used by call media, else clean up media transports starting
* from index idx that have been removed by remote.
/* Clean up media transports in provisional media that is not used by
* call media.
*/
static void media_prov_clean_up(pjsua_call_id call_id, int idx)
void pjsua_media_prov_clean_up(pjsua_call_id call_id)
{
pjsua_call *call = &pjsua_var.calls[call_id];
unsigned i;
if (idx > 0 || call->med_prov_cnt > call->med_cnt) {
if (call->med_prov_cnt > call->med_cnt) {
PJ_LOG(4,(THIS_FILE, "Call %d: cleaning up provisional media, "
"prov_med_cnt=%d, med_cnt=%d",
call_id, (idx == 0? call->med_prov_cnt: idx),
call->med_cnt));
call_id, call->med_prov_cnt, call->med_cnt));
}
for (i = idx; i < call->med_prov_cnt; ++i) {
for (i = 0; i < call->med_prov_cnt; ++i) {
pjsua_call_media *call_med = &call->media_prov[i];
unsigned j;
pj_bool_t used = PJ_FALSE;
@ -1553,7 +1617,7 @@ static void media_prov_clean_up(pjsua_call_id call_id, int idx)
continue;
for (j = 0; j < call->med_cnt; ++j) {
if (idx == 0 && call->media[j].tp == call_med->tp) {
if (call->media[j].tp == call_med->tp) {
used = PJ_TRUE;
break;
}
@ -1573,11 +1637,6 @@ static void media_prov_clean_up(pjsua_call_id call_id, int idx)
call->med_prov_cnt = 0;
}
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,
@ -1664,6 +1723,10 @@ pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
/* Get media count for each media type */
if (rem_sdp) {
/* We are sending answer, check media count for each media type
* from the remote SDP.
*/
sort_media(rem_sdp, &STR_AUDIO, acc->cfg.use_srtp,
maudidx, &maudcnt, &mtotaudcnt);
if (maudcnt==0) {
@ -1695,22 +1758,22 @@ pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
} else {
/* If call already established, calculate media count from current
* local active SDP and call setting. Otherwise, calculate media
* count from the call setting only.
/* If call is already established, adjust the existing call media list
* to media count setting in call setting, e.g: re-enable/disable/add
* media from existing media.
* Otherwise, apply media count from the call setting directly.
*/
if (reinit) {
const pjmedia_sdp_session *sdp;
status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &sdp);
pj_assert(status == PJ_SUCCESS);
sort_media(sdp, &STR_AUDIO, acc->cfg.use_srtp,
maudidx, &maudcnt, &mtotaudcnt);
/* We are sending reoffer, check media count for each media type
* from the existing call media list.
*/
sort_media2(call->media_prov, call->med_prov_cnt,
PJMEDIA_TYPE_AUDIO, maudidx, &maudcnt, &mtotaudcnt);
pj_assert(maudcnt > 0);
sort_media(sdp, &STR_VIDEO, acc->cfg.use_srtp,
mvididx, &mvidcnt, &mtotvidcnt);
sort_media2(call->media_prov, call->med_prov_cnt,
PJMEDIA_TYPE_VIDEO, mvididx, &mvidcnt, &mtotvidcnt);
/* Call setting may add or remove media. Adding media is done by
* enabling any disabled/port-zeroed media first, then adding new
@ -1855,8 +1918,9 @@ pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_DISABLED);
}
/* Put media type just for info */
call_med->type = media_type;
/* Put media type just for info if not yet defined */
if (call_med->type == PJMEDIA_TYPE_NONE)
call_med->type = media_type;
}
}
@ -1991,11 +2055,12 @@ pj_status_t pjsua_media_channel_create_sdp(pjsua_call_id call_id,
if (rem_sdp && mi >= rem_sdp->media_count) {
/* Remote might have removed some media lines. */
for (i = rem_sdp->media_count; i < call->med_prov_cnt; ++i) {
stop_media_stream(call, i);
}
media_prov_clean_up(call->index, rem_sdp->media_count);
call->med_prov_cnt = rem_sdp->media_count;
/* Note that we must not modify the current active media
* (e.g: stop stream, close/cleanup media transport), as if
* SDP nego fails, the current active media should be maintained.
* Also note that our media count should never decrease, even when
* remote removed some media lines.
*/
break;
}
@ -2989,7 +3054,9 @@ on_check_med_status:
pj_memcpy(call->media, call->media_prov,
sizeof(call->media_prov[0]) * call->med_prov_cnt);
/* Perform SDP re-negotiation if needed. */
/* Perform SDP re-negotiation if some media have just got disabled
* in this function due to media count limit settings.
*/
if (got_media && need_renego_sdp) {
pjmedia_sdp_neg *neg = call->inv->neg;

7
tests/pjsua/scripts-sipp/uac-ticket-1864-scenario1.py Normal file
View File

@ -0,0 +1,7 @@
# $Id$
#
import inc_const as const
PJSUA = ["--null-audio --max-calls=1 --auto-answer=200"]
PJSUA_EXPECTS = []

226
tests/pjsua/scripts-sipp/uac-ticket-1864-scenario1.xml Normal file
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@ -0,0 +1,226 @@
<?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 -->
<!-- -->
<!-- -->
<scenario name="Ticket #1864 scenario 1">
<!-- UAC -->
<!-- offer with two media lines -->
<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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
s=3cxPS Audio call
c=IN IP4 172.20.73.100
t=0 0
m=audio 7016 RTP/SAVP 0
c=IN IP4 172.20.73.100
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MFtlKkxOZF16aGBwK0Veaks2JC5+cDBnLWlyeUhN
m=audio 7016 RTP/AVP 0
c=IN IP4 172.20.73.100
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="200" rtd="true">
</recv>
<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=[call_number]
To: sut <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
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
]]>
</send>
<pause milliseconds="1000"/>
<!-- reoffer with one media line -->
<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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
s=3cxPS Audio call
c=IN IP4 172.20.73.100
t=0 0
m=audio 7016 RTP/AVP 0
c=IN IP4 172.20.73.100
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="200" rtd="true">
</recv>
<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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
]]>
</send>
<pause milliseconds="1000"/>
<!-- another offer with two media lines -->
<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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
s=3cxPS Audio call
c=IN IP4 172.20.73.100
t=0 0
m=audio 7016 RTP/SAVP 0
c=IN IP4 172.20.73.100
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MFtlKkxOZF16aGBwK0Veaks2JC5+cDBnLWlyeUhN
m=audio 7016 RTP/AVP 0
c=IN IP4 172.20.73.100
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="200" rtd="true">
</recv>
<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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 ACK
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
]]>
</send>
<pause milliseconds="1000"/>
<send>
<![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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 4 BYE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
]]>
</send>
<recv response="200" rtd="true">
</recv>
<!-- 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>

7
tests/pjsua/scripts-sipp/uac-ticket-1864-scenario2.py Normal file
View File

@ -0,0 +1,7 @@
# $Id$
#
import inc_const as const
PJSUA = ["--null-audio --max-calls=1 --dis-codec=pcma --auto-answer=200"]
PJSUA_EXPECTS = []

227
tests/pjsua/scripts-sipp/uac-ticket-1864-scenario2.xml Normal file
View File

@ -0,0 +1,227 @@
<?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 -->
<!-- -->
<!-- -->
<scenario name="Ticket #1864 scenario 2">
<!-- UAC -->
<!-- offer with two media lines -->
<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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
s=3cxPS Audio call
c=IN IP4 172.20.73.100
t=0 0
m=audio 7016 RTP/SAVP 0
c=IN IP4 172.20.73.100
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MFtlKkxOZF16aGBwK0Veaks2JC5+cDBnLWlyeUhN
m=audio 7016 RTP/AVP 0
c=IN IP4 172.20.73.100
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="200" rtd="true">
</recv>
<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=[call_number]
To: sut <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
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
]]>
</send>
<pause milliseconds="1000"/>
<!-- reoffer with one media line and remote unsupported codec -->
<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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
s=3cxPS Audio call
c=IN IP4 172.20.73.100
t=0 0
m=audio 7016 RTP/AVP 8
c=IN IP4 172.20.73.100
]]>
</send>
<recv response="100" optional="true">
</recv>
<!-- remote rejects the offer due to no matching codec -->
<recv response="500" rtd="true">
</recv>
<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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
]]>
</send>
<pause milliseconds="1000"/>
<!-- reoffer with two media lines and supported codec -->
<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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=3cxPS 518818627584 526334820353 IN IP4 172.20.73.100
s=3cxPS Audio call
c=IN IP4 172.20.73.100
t=0 0
m=audio 7016 RTP/SAVP 0
c=IN IP4 172.20.73.100
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:MFtlKkxOZF16aGBwK0Veaks2JC5+cDBnLWlyeUhN
m=audio 7016 RTP/AVP 0
c=IN IP4 172.20.73.100
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="200" rtd="true">
</recv>
<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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 3 ACK
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
]]>
</send>
<pause milliseconds="1000"/>
<send>
<![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=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 4 BYE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
]]>
</send>
<recv response="200" rtd="true">
</recv>
<!-- 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>