From: Terry Wilson Date: Wed, 2 Nov 2011 22:59:36 +0000 (+0000) Subject: Remove registertrying option in chan_sip X-Git-Tag: 1.8.9.0-rc1~107 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=5844133aba405eadd6ed76516596a606e74baa15;p=thirdparty%2Fasterisk.git Remove registertrying option in chan_sip This option is not only useless, but has been broken since inception since the flag was never copied from the peer where it is set to the pvt where it was checked. RFC 3261 specificially states that you should not send a provisional response to a non-INVITE request, and if we did fix the code so that it worked, it would cause the same kind of user enumeration vulnerability that we've discussed with the nat= setting. This patch removes registertrying option and any code that would have sent a 100 response to a register. Review: https://reviewboard.asterisk.org/r/1562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343220 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 229f035f8b..df549512af 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -14411,8 +14411,6 @@ static enum check_auth_result register_verify(struct sip_pvt *p, struct ast_sock res = AUTH_PEER_NOT_DYNAMIC; } else { ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_NAT_FORCE_RPORT); - if (ast_test_flag(&p->flags[1], SIP_PAGE2_REGISTERTRYING)) - transmit_response(p, "100 Trying", req); if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri2, XMIT_UNRELIABLE, req->ignore))) { if (sip_cancel_destroy(p)) ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); @@ -14502,14 +14500,6 @@ static enum check_auth_result register_verify(struct sip_pvt *p, struct ast_sock ao2_unlock(peer); } } - if (!peer && sip_cfg.alwaysauthreject && ast_test_flag(&p->flags[1], SIP_PAGE2_REGISTERTRYING)) { - /* If we found a peer, we transmit a 100 Trying. Therefore, if we're - * trying to avoid leaking information, we MUST also transmit the same - * response when we DON'T find a peer. */ - transmit_response(p, "100 Trying", req); - /* Insert a fake delay between the 100 and the subsequent failure. */ - sched_yield(); - } if (!res) { ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name); } @@ -17079,7 +17069,6 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct ast_cli(fd, ")\n"); ast_cli(fd, " Auto-Framing : %s \n", AST_CLI_YESNO(peer->autoframing)); - ast_cli(fd, " 100 on REG : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_REGISTERTRYING))); ast_cli(fd, " Status : "); peer_status(peer, status, sizeof(status)); ast_cli(fd, "%s\n", status); @@ -27014,8 +27003,6 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str } } else if (!strcasecmp(v->name, "preferred_codec_only")) { ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_PREFERRED_CODEC); - } else if (!strcasecmp(v->name, "registertrying")) { - ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_REGISTERTRYING); } else if (!strcasecmp(v->name, "autoframing")) { peer->autoframing = ast_true(v->value); } else if (!strcasecmp(v->name, "rtptimeout")) { diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h index a3f55cf7c9..aa6cfcd3ab 100644 --- a/channels/sip/include/sip.h +++ b/channels/sip/include/sip.h @@ -336,11 +336,10 @@ #define SIP_PAGE2_FAX_DETECT_T38 (2 << 23) /*!< DP: Fax Detection support - detect T.38 reinvite from peer */ #define SIP_PAGE2_FAX_DETECT_BOTH (3 << 23) /*!< DP: Fax Detection support - detect both */ -#define SIP_PAGE2_REGISTERTRYING (1 << 24) /*!< DP: Send 100 Trying on REGISTER attempts */ -#define SIP_PAGE2_UDPTL_DESTINATION (1 << 25) /*!< DP: Use source IP of RTP as destination if NAT is enabled */ -#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 26) /*!< DP: Always set up video, even if endpoints don't support it */ -#define SIP_PAGE2_HAVEPEERCONTEXT (1 << 27) /*< Are we associated with a configured peer context? */ -#define SIP_PAGE2_USE_SRTP (1 << 28) /*!< DP: Whether we should offer (only) SRTP */ +#define SIP_PAGE2_UDPTL_DESTINATION (1 << 24) /*!< DP: Use source IP of RTP as destination if NAT is enabled */ +#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 25) /*!< DP: Always set up video, even if endpoints don't support it */ +#define SIP_PAGE2_HAVEPEERCONTEXT (1 << 26) /*< Are we associated with a configured peer context? */ +#define SIP_PAGE2_USE_SRTP (1 << 27) /*!< DP: Whether we should offer (only) SRTP */ #define SIP_PAGE2_FLAGS_TO_COPY \ (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \ diff --git a/contrib/realtime/mysql/sippeers.sql b/contrib/realtime/mysql/sippeers.sql index 2393dd4763..6884ac6a9e 100644 --- a/contrib/realtime/mysql/sippeers.sql +++ b/contrib/realtime/mysql/sippeers.sql @@ -62,7 +62,6 @@ CREATE TABLE IF NOT EXISTS `sippeers` ( `sendrpid` enum('yes','no') DEFAULT NULL, `outboundproxy` varchar(40) DEFAULT NULL, `callbackextension` varchar(40) DEFAULT NULL, - `registertrying` enum('yes','no') DEFAULT NULL, `timert1` int(11) DEFAULT NULL, `timerb` int(11) DEFAULT NULL, `qualifyfreq` int(11) DEFAULT NULL,