From: Paul Cadach <paul@odt.east.telecom.kz>
Date: Sat, 30 Sep 2006 04:37:39 +0000 (+0000)
Subject: Found some buggy SIP clients (phones Planet VIP-153T firmware
X-Git-Tag: 1.4.0-beta3~113
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=7a7c5011a1052d84a2c7b853f4bb935d49efcc9a;p=thirdparty%2Fasterisk.git

Found some buggy SIP clients (phones Planet VIP-153T firmware
1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK not on OK
message only (when remote party answers) but on RINGING message
too, so when we send 200 OK message, we get unidentified ACK
message (because INVITE acknowledged on RINGING message already),
so 200 OK retransmits within its retransmission interval then
call gets dropped.

If someone else knows how to provide workaround for such cases,
please, fix it in correct way.

Thanks to ssh from #asteriskru for provide access to his box to
study and fix this case.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 7030741e8c..79b9ecfe46 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -6109,6 +6109,8 @@ static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct s
 		add_t38_sdp(&resp, p);
 	} else 
 		ast_log(LOG_ERROR, "Can't add SDP to response, since we have no UDPTL session allocated. Call-ID %s\n", p->callid);
+	if (retrans && !p->pendinginvite)
+		p->pendinginvite = seqno;		/* Buggy clients sends ACK on RINGING too */
 	return send_response(p, &resp, retrans, seqno);
 }
 
@@ -6147,6 +6149,8 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const
 		add_sdp(&resp, p);
 	} else 
 		ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
+	if (reliable && !p->pendinginvite)
+		p->pendinginvite = seqno;		/* Buggy clients sends ACK on RINGING too */
 	return send_response(p, &resp, reliable, seqno);
 }