diff options
| author | pcadach <pcadach@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-09-30 04:41:04 +0000 |
|---|---|---|
| committer | pcadach <pcadach@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-09-30 04:41:04 +0000 |
| commit | daebff01d175fd436ddf64a7435d69c40ecf7d8d (patch) | |
| tree | f2dd643dc80cc1ee6f009d146f404842aee2f340 /channels | |
| parent | 8f0144e93d26bdbbd2dd12617ba468c76effaa0e (diff) | |
Merged revisions 44068 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r44068 | pcadach | 2006-09-30 10:37:39 +0600 (Сбт, 30 Сен 2006) | 14 lines
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: http://svn.digium.com/svn/asterisk/trunk@44069 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 4 |
1 files changed, 4 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 99edcb10b..ec45fc024 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -6100,6 +6100,8 @@ 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); } @@ -6138,6 +6140,8 @@ 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); } |