diff options
| author | Jonathan Rose <jrose@digium.com> | 2013-09-27 17:46:16 +0000 |
|---|---|---|
| committer | Jonathan Rose <jrose@digium.com> | 2013-09-27 17:46:16 +0000 |
| commit | 7e2a72771dbd538d8c72865406ca5aabb883d9d9 (patch) | |
| tree | 6958867c3df6be5b0ced0d64592c77715f8d9b93 | |
| parent | 7c796593d347210ec8da2ba80b4483d261ef8f05 (diff) | |
chan_sip: Reject calls on 200 OKs if no SDP has been received
When Asterisk receives a 200 OK in response to an invite, that peer should have
sent an SDP at some point by then. If the channel has never received an SDP,
media won't have been set and the remote address won't be known. Endpoints in
general should not be doing this. This patch makes it so that Asterisk will
simply hang up a call if it sends a 200 OK at this point. So far this odd
behavior for endpoints has only been observed in tests which involved manually
created SIP transactions in SIPp.
(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/
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Merged revisions 399939 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 399962 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 399976 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399978 65c4cc65-6c06-0410-ace0-fbb531ad65f3
| -rw-r--r-- | channels/chan_sip.c | 13 | ||||
| -rw-r--r-- | channels/sip/include/sip.h | 1 |
2 files changed, 14 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index aaae83c6f..17083c6a0 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -23250,6 +23250,15 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest ast_set_flag(&p->flags[0], SIP_PENDINGBYE); } ast_rtp_instance_activate(p->rtp); + } else if (!reinvite) { + struct ast_sockaddr remote_address = {{0,}}; + + ast_rtp_instance_get_remote_address(p->rtp, &remote_address); + if (ast_sockaddr_isnull(&remote_address) || (!ast_strlen_zero(p->theirprovtag) && strcmp(p->theirtag, p->theirprovtag))) { + ast_log(LOG_WARNING, "Received response: \"200 OK\" from '%s' without SDP\n", p->relatedpeer->name); + ast_set_flag(&p->flags[0], SIP_PENDINGBYE); + ast_rtp_instance_activate(p->rtp); + } } if (!req->ignore && p->owner) { @@ -24200,7 +24209,11 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc gettag(req, "To", tag, sizeof(tag)); ast_string_field_set(p, theirtag, tag); + } else { + /* Store theirtag to track for changes when 200 responses to invites are received without SDP */ + ast_string_field_set(p, theirprovtag, p->theirtag); } + /* This needs to be configurable on a channel/peer level, not mandatory for all communication. Sadly enough, NAT implementations are not so stable so we can always rely on these headers. diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h index c0704ec79..342c87b36 100644 --- a/channels/sip/include/sip.h +++ b/channels/sip/include/sip.h @@ -1019,6 +1019,7 @@ struct sip_pvt { AST_STRING_FIELD(rdnis); /*!< Referring DNIS */ AST_STRING_FIELD(redircause); /*!< Referring cause */ AST_STRING_FIELD(theirtag); /*!< Their tag */ + AST_STRING_FIELD(theirprovtag); /*!< Provisional their tag, used when evaluating responses to invites */ AST_STRING_FIELD(tag); /*!< Our tag for this session */ AST_STRING_FIELD(username); /*!< [user] name */ AST_STRING_FIELD(peername); /*!< [peer] name, not set if [user] */ |