Merged revisions 165180,200689 via svnmerge from

https://origsvn.digium.com/svn/asterisk/trunk

........
  r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14 lines
  
  This patch adds a new 'ignoresdpversion' option to sip.conf.  When this is
  enabled (either globally or for a specific peer), chan_sip will treat any SDP
  data it receives as new data and update the media stream accordingly.  By
  default, Asterisk will only modify the media stream if the SDP session version
  received is different from the current SDP session version.  This option is
  required to interoperate with devices that have non-standard SDP session
  version implementations (observed by toc on the bug tracker with Microsoft OCS
  which always uses 0 as the session version).
  
  http://reviewboard.digium.com/r/94/
  (closes issue #13958)
  Reported by: toc
  Tested by: toc
........
  r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12 lines
  
  Accept T.38 re-INVITE responses with invalid SDP versions.
  
  This commit changes the 'incoming SDP version' check logic a bit more; when
  'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to
  switch to T.38, we'll always accept the peer's SDP response, even if they
  don't properly increment the SDP version number as they should. If this situation
  occurs, a warning message will be generated suggesting that the peer's
  configuration be changed to include the 'ignoresdpversion' configuration option
  (although ideally they'd fix their SIP implementation to be RFC compliant).
  
  AST-221
........


git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@200724 f38db490-d61c-443f-a65b-d21fe96a405b

2 files changed, 65 insertions, 21 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index a807d1a62..6a919ef6c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -981,6 +981,7 @@ struct sip_auth {
 #define SIP_PAGE2_ALLOWSUBSCRIBE	(1 << 16)	/*!< GP: Allow subscriptions from this peer? */
 #define SIP_PAGE2_ALLOWOVERLAP		(1 << 17)	/*!< DP: Allow overlap dialing ? */
 #define SIP_PAGE2_SUBSCRIBEMWIONLY	(1 << 18)	/*!< GP: Only issue MWI notification if subscribed to */
+#define SIP_PAGE2_IGNORESDPVERSION	(1 << 19)	/*!< GDP: Ignore the SDP session version number we receive and treat all sessions as new */
 
 #define SIP_PAGE2_T38SUPPORT		(7 << 20)	/*!< GDP: T38 Fax Passthrough Support */
 #define SIP_PAGE2_T38SUPPORT_UDPTL	(1 << 20)	/*!< GDP: T38 Fax Passthrough Support */
@@ -999,9 +1000,9 @@ struct sip_auth {
 #define SIP_PAGE2_UDPTL_DESTINATION     (1 << 30)       /*!< DP: Use source IP of RTP as destination if NAT is enabled */
 
 #define SIP_PAGE2_FLAGS_TO_COPY \
-	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
-	SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
-	SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_UDPTL_DESTINATION)
+	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
+	SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
+	 SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_UDPTL_DESTINATION)
 
 /*@}*/ 
 
@@ -6761,13 +6762,39 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 		return -1;
 	}
 
-	if (p->sessionversion_remote < 0 || p->sessionversion_remote != rua_version) {
- 		p->sessionversion_remote = rua_version;
+	/* we need to check the SDP version number the other end sent us;
+	 * our rules for deciding what to accept are a bit complex.
+	 *
+	 * 1) if 'ignoresdpversion' has been set for this dialog, then
+	 *    we will just accept whatever they sent and assume it is
+	 *    a modification of the session, even if it is not
+	 * 2) otherwise, if this is the first SDP we've seen from them
+	 *    we accept it
+	 * 3) otherwise, if the new SDP version number is higher than the
+	 *    old one, we accept it
+	 * 4) otherwise, if this SDP is in response to us requesting a switch
+	 *    to T.38, we accept the SDP, but also generate a warning message
+	 *    that this peer should have the 'ignoresdpversion' option set,
+	 *    because it is not following the SDP offer/answer RFC; if we did
+	 *    not request a switch to T.38, then we stop parsing the SDP, as it
+	 *    has not changed from the previous version
+	 */
+
+	if (ast_test_flag(&p->flags[1], SIP_PAGE2_IGNORESDPVERSION) ||
+	    (p->sessionversion_remote < 0) ||
+	    (p->sessionversion_remote < rua_version)) {
+		p->sessionversion_remote = rua_version;
 		p->session_modify = TRUE;
-	} else if (p->sessionversion_remote == rua_version) {
-		p->session_modify = FALSE;
-		ast_debug(2, "SDP version number same as previous SDP\n");
-		return 0;
+	} else {
+		if (p->t38.state == T38_LOCAL_REINVITE) {
+			p->sessionversion_remote = rua_version;
+			p->session_modify = TRUE;
+			ast_log(LOG_WARNING, "Call %s responded to our T.38 reinvite without changing SDP version; 'ignoresdpversion' should be set for this peer.\n", p->callid);
+		} else {
+			p->session_modify = FALSE;
+			ast_debug(2, "Call %s responded to our reinvite without changing SDP version; ignoring SDP.\n", p->callid);
+			return 0;
+		}
 	} 
 
 	/* Try to find first media stream */
@@ -13237,6 +13264,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
 		ast_cli(fd, "  User=Phone   : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)));
 		ast_cli(fd, "  Video Support: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)));
 		ast_cli(fd, "  Text Support : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)));
+		ast_cli(fd, "  Ign SDP ver  : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_IGNORESDPVERSION)));
 		ast_cli(fd, "  Trust RPID   : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_TRUSTRPID)));
 		ast_cli(fd, "  Send RPID    : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_SENDRPID)));
 		ast_cli(fd, "  Subscriptions: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
@@ -13439,6 +13467,7 @@ static char *sip_show_user(struct ast_cli_entry *e, int cmd, struct ast_cli_args
  		ast_cli(a->fd, "  Sess-Refresh : %s\n", strefresher2str(user->stimer.st_ref));
  		ast_cli(a->fd, "  Sess-Expires : %d secs\n", user->stimer.st_max_se);
  		ast_cli(a->fd, "  Sess-Min-SE  : %d secs\n", user->stimer.st_min_se);
+		ast_cli(a->fd, "  Ign SDP ver  : %s\n", cli_yesno(ast_test_flag(&user->flags[1], SIP_PAGE2_IGNORESDPVERSION)));
 
 		ast_cli(a->fd, "  Codec Order  : (");
 		print_codec_to_cli(a->fd, &user->prefs);
@@ -13590,6 +13619,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
 	ast_cli(a->fd, "  Videosupport:           %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT)));
 	ast_cli(a->fd, "  Textsupport:            %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT)));
 	ast_cli(a->fd, "  AutoCreate Peer:        %s\n", cli_yesno(autocreatepeer));
+	ast_cli(a->fd, "  Ignore SDP sess. ver.:  %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_IGNORESDPVERSION)));
 	ast_cli(a->fd, "  Match Auth Username:    %s\n", cli_yesno(global_match_auth_username));
 	ast_cli(a->fd, "  Allow unknown access:   %s\n", cli_yesno(global_allowguest));
 	ast_cli(a->fd, "  Allow subscriptions:    %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
@@ -20496,6 +20526,9 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
 	} else if (!strcasecmp(v->name, "allowsubscribe")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_ALLOWSUBSCRIBE);
 		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWSUBSCRIBE);
+	} else if (!strcasecmp(v->name, "ignoresdpversion")) {
+		ast_set_flag(&mask[1], SIP_PAGE2_IGNORESDPVERSION);
+		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_IGNORESDPVERSION);
 	} else if (!strcasecmp(v->name, "t38pt_udptl")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_UDPTL);
 		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_UDPTL);
@@ -21613,6 +21646,7 @@ static int reload_config(enum channelreloadreason reason)
 
 	ast_clear_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT);
 	ast_clear_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT);
+	ast_clear_flag(&global_flags[1], SIP_PAGE2_IGNORESDPVERSION);
 
 	/* Read the [general] config section of sip.conf (or from realtime config) */
 	for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 0fb6afe54..a769f189e 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -570,6 +570,15 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                 ; instead of INVITE. This can be combined with 'nonat', as
                                 ; 'canreinvite=update,nonat'. It implies 'yes'.
 
+;ignoresdpversion=yes           ; By default, Asterisk will honor the session version
+                                ; number in SDP packets and will only modify the SDP
+                                ; session if the version number changes. This option will
+                                ; force asterisk to ignore the SDP session version number
+                                ; and treat all SDP data as new data.  This is required
+                                ; for devices that send us non standard SDP packets
+                                ; (observed with Microsoft OCS). By default this option is
+                                ; off.
+
 ;----------------------------------------- REALTIME SUPPORT ------------------------
 ; For additional information on ARA, the Asterisk Realtime Architecture,
 ; please read realtime.txt and extconfig.txt in the /doc directory of the
@@ -729,12 +738,19 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ; subscribecontext            subscribecontext
 ; videosupport                videosupport
 ; maxcallbitrate              maxcallbitrate
-; rfc2833compensate           mailbox
-; session-timers              busylevel
-; session-expires            
-; session-minse               template
-; session-refresher           fromdomain
-; t38pt_usertpsource          regexten
+; rfc2833compensate           rfc2833compensate
+; ignoresdpversion            ignoresdpversion
+; session-timers              session-timers
+; session-expires             session-expires
+; session-minse               session-minse
+; session-refresher           session-refresher
+; t38pt_usertpsource          t38pt_usertpsource
+;                             regexten
+;                             template
+;                             fromdomain
+;                             regexten
+;                             mailbox
+;                             busylevel
 ;                             fromuser
 ;                             host
 ;                             port
@@ -745,17 +761,11 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;                             rtpholdtimeout
 ;                             sendrpid
 ;                             outboundproxy
-;                             rfc2833compensate
 ;                             callbackextension
 ;                             registertrying
-;                             session-timers
-;                             session-expires
-;                             session-minse
-;                             session-refresher
 ;                             timert1
 ;                             timerb
 ;                             qualifyfreq
-;                             t38pt_usertpsource
 ;                             contactpermit         ; Limit what a host may register as (a neat trick
 ;                             contactdeny           ; is to register at the same IP as a SIP provider,
 ;                                                   ; then call oneself, and get redirected to that