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authorRussell Bryant <russell@russellbryant.com>2008-05-08 19:17:04 +0000
committerRussell Bryant <russell@russellbryant.com>2008-05-08 19:17:04 +0000
commitc961d9637f5d64a85b5990311c90112a5b2a4a40 (patch)
tree56bc8463b99e99394f8033bf64b0e3fa6aec391c /channels
parentc02cf176e134ea05a705a6a7ca6cf27317bcf1cd (diff)
Merged revisions 115565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115565 | russell | 2008-05-08 14:15:25 -0500 (Thu, 08 May 2008) | 33 lines Merged revisions 115564 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) | 25 lines Fix a race condition that bbryant just found while doing some IAX2 testing. He was running Asterisk trunk running IAX2 calls through a few Asterisk boxes, however, the audio was extremely choppy. We looked at a packet trace and saw a storm of INVAL and VNAK frames being sent from one box to another. It turned out that what had happened was that one box tried to send a CONTROL frame before the 3 way handshake had completed. So, that frame did not include the destination call number, because it didn't have it yet. Part of our recent work for security issues included an additional check to ensure that frames that are supposed to include the destination call number have the correct one. This caused the frame to be rejected with an INVAL. The frame would get retransmitted for forever, rejected every time ... This race condition exists in all versions that got the security changes, in theory. However, it is really only likely that this would cause a problem in Asterisk trunk. There was a control frame being sent (SRCUPDATE) at the _very_ beginning of the call, which does not exist in 1.2 or 1.4. However, I am fixing all versions that could potentially be affected by the introduced race condition. These changes are what bbryant and I came up with to fix the issue. Instead of simply dropping control frames that get sent before the handshake is complete, the code attempts to wait a little while, since in most cases, the handshake will complete very quickly. If it doesn't complete after yielding for a little while, then the frame gets dropped. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115566 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_iax2.c18
1 files changed, 18 insertions, 0 deletions
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index 5bdea6592..73eb3f138 100644
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -3927,6 +3927,24 @@ static int iax2_indicate(struct ast_channel *c, int condition, const void *data,
ast_mutex_lock(&iaxsl[callno]);
pvt = iaxs[callno];
+ if (!pvt->peercallno) {
+ /* We don't know the remote side's call number, yet. :( */
+ int count = 10;
+ while (count-- && pvt && !pvt->peercallno) {
+ ast_mutex_unlock(&iaxsl[callno]);
+ usleep(1);
+ ast_mutex_lock(&iaxsl[callno]);
+ pvt = iaxs[callno];
+ }
+ if (pvt->peercallno) {
+ ast_log(LOG_NOTICE, "Yay, we didn't know the peercallno, but we were patient and got it.\n");
+ } else {
+ ast_log(LOG_NOTICE, "Damnit! We waited around and never got the peercallno ...\n");
+ res = -1;
+ goto done;
+ }
+ }
+
switch (condition) {
case AST_CONTROL_HOLD:
if (strcasecmp(pvt->mohinterpret, "passthrough")) {