[svn-commits] oej: branch oej/darjeeling-prack-1.8 r369790 - /team/oej/darjeeling-prack-1.8...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Jul 9 06:24:31 CDT 2012
Author: oej
Date: Mon Jul 9 06:24:29 2012
New Revision: 369790
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369790
Log:
Update patch.
Modified:
team/oej/darjeeling-prack-1.8/patches/darjeeling-prack-1.8.diff
Modified: team/oej/darjeeling-prack-1.8/patches/darjeeling-prack-1.8.diff
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-1.8/patches/darjeeling-prack-1.8.diff?view=diff&rev=369790&r1=369789&r2=369790
==============================================================================
--- team/oej/darjeeling-prack-1.8/patches/darjeeling-prack-1.8.diff (original)
+++ team/oej/darjeeling-prack-1.8/patches/darjeeling-prack-1.8.diff Mon Jul 9 06:24:29 2012
@@ -1,7 +1,7 @@
Index: channels/chan_sip.c
===================================================================
---- channels/chan_sip.c (.../branches/1.8) (revision 369624)
-+++ channels/chan_sip.c (.../team/oej/darjeeling-prack-1.8) (revision 369624)
+--- channels/chan_sip.c (.../branches/1.8) (revision 369789)
++++ channels/chan_sip.c (.../team/oej/darjeeling-prack-1.8) (revision 369789)
@@ -1242,10 +1242,10 @@
static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
static int retrans_pkt(const void *data);
@@ -66,7 +66,7 @@
}
- ast_debug(1, "Stopping retransmission on '%s' of %s %u: Match %s\n",
- p->callid, resp ? "Response" : "Request", seqno, msg);
-+ ast_debug(1, "Stopping retransmission on '%s' of %s %u: Match %s Rseq %d\n",
++ ast_debug(1, "Stopping retransmission on '%s' of %s %u: Match %s Rseq %u\n",
+ p->callid, resp ? "Response" : "Request", seqno, msg, rseqno);
return res;
}
@@ -144,6 +144,15 @@
finalize_content(req);
add_blank(req);
if (sip_debug_test_pvt(p)) {
+@@ -4187,7 +4248,7 @@
+ }
+
+ res = (reliable) ?
+- __sip_reliable_xmit(p, seqno, 1, req->data, (reliable == XMIT_CRITICAL), req->method) :
++ __sip_reliable_xmit(p, seqno, 1, req->data, (reliable == XMIT_CRITICAL || reliable == XMIT_PRACK), req->method) :
+ __sip_xmit(p, req->data);
+ deinit_req(req);
+ if (res > 0) {
@@ -6485,7 +6546,13 @@
struct sip_pvt *p = ast->tech_pvt;
@@ -159,7 +168,7 @@
try_suggested_sip_codec(p);
ast_setstate(ast, AST_STATE_UP);
-@@ -9871,13 +9938,15 @@
+@@ -9873,13 +9940,15 @@
* is supported for this dialog. */
static int add_supported_header(struct sip_pvt *pvt, struct sip_request *req)
{
@@ -180,7 +189,7 @@
}
/*! \brief Add header to SIP message */
-@@ -10313,6 +10382,7 @@
+@@ -10315,6 +10384,7 @@
snprintf(se_hdr, sizeof(se_hdr), "%d;refresher=%s", p->stimer->st_interval,
strefresher2str(p->stimer->st_ref));
add_header(resp, "Session-Expires", se_hdr);
@@ -188,7 +197,7 @@
}
if (msg[0] == '2' && (p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER)) {
-@@ -10497,8 +10567,10 @@
+@@ -10499,8 +10569,10 @@
{
struct sip_request resp;
uint32_t seqno = 0;
@@ -200,7 +209,7 @@
ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq"));
return -1;
}
-@@ -10510,6 +10582,10 @@
+@@ -10512,6 +10584,10 @@
ast_clear_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
add_rpid(&resp, p);
}
@@ -211,7 +220,7 @@
if (ast_test_flag(&p->flags[0], SIP_OFFER_CC)) {
add_cc_call_info_to_response(p, &resp);
}
-@@ -10549,6 +10625,10 @@
+@@ -10551,6 +10627,10 @@
add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
}
}
@@ -222,7 +231,7 @@
return send_response(p, &resp, reliable, seqno);
}
-@@ -10561,6 +10641,7 @@
+@@ -10563,6 +10643,7 @@
}
respprep(&resp, p, msg, req);
add_header(&resp, "SIP-ETag", esc_entry->entity_tag);
@@ -230,7 +239,7 @@
return send_response(p, &resp, 0, 0);
}
-@@ -10648,6 +10729,7 @@
+@@ -10650,6 +10731,7 @@
respprep(&resp, p, msg, req);
append_date(&resp);
add_header(&resp, "Unsupported", unsupported);
@@ -238,7 +247,7 @@
return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
-@@ -10701,6 +10783,7 @@
+@@ -10703,6 +10785,7 @@
struct sip_request resp;
respprep(&resp, p, msg, req);
append_date(&resp);
@@ -246,7 +255,7 @@
return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
-@@ -10710,6 +10793,7 @@
+@@ -10712,6 +10795,7 @@
struct sip_request resp;
respprep(&resp, p, msg, req);
add_header(&resp, "Accept", "application/sdp");
@@ -254,7 +263,7 @@
return send_response(p, &resp, reliable, 0);
}
-@@ -10722,6 +10806,7 @@
+@@ -10724,6 +10808,7 @@
snprintf(tmp, sizeof(tmp), "%d", min_expiry);
respprep(&resp, p, msg, req);
add_header(&resp, "Min-Expires", tmp);
@@ -262,7 +271,7 @@
return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
-@@ -11803,6 +11888,16 @@
+@@ -11805,6 +11890,18 @@
if (rpid == TRUE) {
add_rpid(&resp, p);
}
@@ -271,15 +280,17 @@
+ reliable = XMIT_PRACK;
+ }
+ if (strncmp(msg, "100", 3)) {
-+ /* If we send a resposne WITH sdp we are not allowed to respond before the PRACK is received */
-+ ast_set_flag(&p->flags[2], SIP_PAGE3_INVITE_WAIT_FOR_PRACK);
++ /* If we send a response WITH sdp we are not allowed to respond before the PRACK is received */
++ if (ast_test_flag(&p->flags[2], SIP_PAGE3_100REL)) {
++ ast_set_flag(&p->flags[2], SIP_PAGE3_INVITE_WAIT_FOR_PRACK);
++ }
+ add_prack_respheader(p, &resp, reliable);
+ add_required_respheader(&resp);
+ }
if (ast_test_flag(&p->flags[0], SIP_OFFER_CC)) {
add_cc_call_info_to_response(p, &resp);
}
-@@ -12212,7 +12307,34 @@
+@@ -12214,7 +12311,69 @@
}
/*!
@@ -288,26 +299,61 @@
+ */
+static int transmit_prack(struct sip_pvt *p, uint32_t their_rseq)
+{
-+ if (their_rseq == p->irseq) {
-+ ast_debug(3, "!?!?!?!?!? This is a retransmit of the previous response. %u \n", their_rseq);
-+ /* RFC 3262: In particular, a UAC SHOULD NOT retransmit the PRACK request
-+ when it receives a retransmission of the provisional response being
-+ acknowledged, although doing so does not create a protocol error.*/
-+ return -2; /* Not used by transmit_invite et al */
-+ }
-+ if (p->irseq > 0 && their_rseq != p->irseq + 1) {
-+ ast_debug(3, "!?!?!?!?!? This is a response out of sequence! ignored. %u \n", their_rseq);
-+ /* RFC 3262: if the UAC receives another reliable provisional
-+ response to the same request, and its RSeq value is not one higher
-+ than the value of the sequence number, that response MUST NOT be
-+ acknowledged with a PRACK, and MUST NOT be processed further by the
-+ UAC. An implementation MAY discard the response, or MAY cache the
-+ response in the hopes of receiving the missing responses.
++ int res;
++ int comparerseq = TRUE;
++ uint32_t focus_rseq = p->irseq;
++
++ /* During the early media phase, we could have a situation where we get provisional
++ responses from multiple devices, in separate early dialogs. In this case, this
++ code focuses on the FIRST early media response as the one in focus where we
++ check the rseq sequence numbers for retransmits and act upon them.
++ */
++
++ if (!ast_strlen_zero(p->theirtag_prack) && strcmp(p->theirtag, p->theirtag_prack)) {
++ /* We have already sent a PRACK in this dialog, but to a different device.
++ In this code, we focus on the first response that requires PRACK and do not check
++ the validity of rseq in responses in other early dialogs by controlling
++ the PRACK sequence numbers ordering.
++
++ To be 100% RFC correct, we should have a sip_pvt structure for each early dialog
++ and terminate them if we get a 199 response in that early dialog. these should
++ be organized in a tree-like structure based on the original
++ INVITE callid, cseq and from-tag.
+ */
-+ return -3;
++ comparerseq = FALSE;
++ }
++
++ if (comparerseq) {
++ if (their_rseq == p->irseq) {
++ ast_debug(3, "!?!?!?!?!? This is a retransmit of the previous response. %u \n", their_rseq);
++ /* RFC 3262: In particular, a UAC SHOULD NOT retransmit the PRACK request
++ when it receives a retransmission of the provisional response being
++ acknowledged, although doing so does not create a protocol error.*/
++ return -2; /* Not used by transmit_invite et al */
++ }
++ if (p->irseq > 0 && their_rseq != p->irseq + 1) {
++ ast_debug(3, "!?!?!?!?!? This is a response out of sequence! ignored. %u \n", their_rseq);
++ /* RFC 3262: if the UAC receives another reliable provisional
++ response to the same request, and its RSeq value is not one higher
++ than the value of the sequence number, that response MUST NOT be
++ acknowledged with a PRACK, and MUST NOT be processed further by the
++ UAC. An implementation MAY discard the response, or MAY cache the
++ response in the hopes of receiving the missing responses.
++ */
++ return -3;
++ }
+ }
+ p->irseq = their_rseq;
-+ return transmit_invite(p, SIP_PRACK, 0, 1, NULL);
++ res = transmit_invite(p, SIP_PRACK, 0, 1, NULL);
++
++ if (ast_strlen_zero(p->theirtag_prack)) {
++ p->irseq = their_rseq;
++ ast_string_field_set(p, theirtag_prack, p->tag); /* Save this tag as a PRACK focus for this dialog */
++ } else {
++ p->irseq = focus_rseq;
++ }
++
++ return res;
+}
+
+/*!
@@ -315,7 +361,7 @@
* \param p sip_pvt structure
* \param sipmethod
* \param sdp unknown
-@@ -12228,7 +12350,9 @@
+@@ -12230,7 +12389,9 @@
req.method = sipmethod;
if (init) {/* Bump branch even on initial requests */
p->branch ^= ast_random();
@@ -326,7 +372,7 @@
build_via(p);
}
if (init > 1) {
-@@ -12264,17 +12388,24 @@
+@@ -12266,17 +12427,24 @@
}
snprintf(buf, sizeof(buf), "%d", p->expiry);
add_header(&req, "Expires", buf);
@@ -352,7 +398,7 @@
char i2astr[10];
if (!p->stimer->st_interval) {
-@@ -17602,6 +17733,7 @@
+@@ -17604,6 +17772,7 @@
ast_cli(fd, " DirectMedia : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)));
ast_cli(fd, " PromiscRedir : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)));
ast_cli(fd, " User=Phone : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)));
@@ -360,7 +406,7 @@
ast_cli(fd, " Video Support: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) || ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS)));
ast_cli(fd, " Text Support : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)));
ast_cli(fd, " Ign SDP ver : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_IGNORESDPVERSION)));
-@@ -17725,6 +17857,7 @@
+@@ -17727,6 +17896,7 @@
/* - is enumerated */
astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
@@ -368,15 +414,17 @@
astman_append(s, "ToHost: %s\r\n", peer->tohost);
astman_append(s, "Address-IP: %s\r\nAddress-Port: %d\r\n", ast_sockaddr_stringify_addr(&peer->addr), ast_sockaddr_port(&peer->addr));
astman_append(s, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_sockaddr_stringify_addr(&peer->defaddr), ast_sockaddr_port(&peer->defaddr));
-@@ -18308,6 +18441,7 @@
+@@ -18309,7 +18479,9 @@
+ ast_cli(a->fd, " Timer T1 minimum: %d\n", global_t1min);
ast_cli(a->fd, " Timer B: %d\n", global_timer_b);
ast_cli(a->fd, " No premature media: %s\n", AST_CLI_YESNO(global_prematuremediafilter));
++ ast_cli(a->fd, " Early media focus: %s\n", AST_CLI_YESNO(sip_cfg.early_media_focus));
ast_cli(a->fd, " Max forwards: %d\n", sip_cfg.default_max_forwards);
+ ast_cli(a->fd, " PRACK support: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[2], SIP_PAGE3_PRACK)));
ast_cli(a->fd, "\nDefault Settings:\n");
ast_cli(a->fd, "-----------------\n");
-@@ -18679,10 +18813,14 @@
+@@ -18681,10 +18853,14 @@
ast_cli(a->fd, " Format: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0) );
ast_cli(a->fd, " T.38 support %s\n", AST_CLI_YESNO(cur->udptl != NULL));
ast_cli(a->fd, " Video support %s\n", AST_CLI_YESNO(cur->vrtp != NULL));
@@ -391,7 +439,7 @@
ast_cli(a->fd, " Force rport: %s\n", AST_CLI_YESNO(ast_test_flag(&cur->flags[0], SIP_NAT_FORCE_RPORT)));
if (ast_sockaddr_isnull(&cur->redirip)) {
ast_cli(a->fd,
-@@ -19928,6 +20066,48 @@
+@@ -19933,6 +20109,48 @@
return 0;
}
@@ -440,7 +488,45 @@
/*!
* \brief Handle authentication challenge for SIP UPDATE
*
-@@ -20366,6 +20546,10 @@
+@@ -20188,7 +20406,7 @@
+ ast_setstate(p->owner, AST_STATE_RINGING);
+ }
+ }
+- if (find_sdp(req)) {
++ if (!req->ignoresdp && find_sdp(req)) {
+ if (p->invitestate != INV_CANCELLED)
+ p->invitestate = INV_EARLY_MEDIA;
+ res = process_sdp(p, req, SDP_T38_NONE);
+@@ -20197,6 +20415,9 @@
+ ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
+ }
+ ast_rtp_instance_activate(p->rtp);
++ if (sip_cfg.early_media_focus && ast_strlen_zero(p->theirtag_early)) {
++ ast_string_field_set(p, theirtag_early, p->tag);
++ }
+ }
+ check_pendings(p);
+ break;
+@@ -20260,7 +20481,7 @@
+ }
+ sip_handle_cc(p, req, AST_CC_CCNR);
+ }
+- if (find_sdp(req)) {
++ if (!req->ignoresdp && find_sdp(req)) {
+ if (p->invitestate != INV_CANCELLED)
+ p->invitestate = INV_EARLY_MEDIA;
+ res = process_sdp(p, req, SDP_T38_NONE);
+@@ -20268,6 +20489,9 @@
+ /* Queue a progress frame */
+ ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
+ }
++ if (sip_cfg.early_media_focus && ast_strlen_zero(p->theirtag_early)) {
++ ast_string_field_set(p, theirtag_early, p->tag);
++ }
+ ast_rtp_instance_activate(p->rtp);
+ } else {
+ /* Alcatel PBXs are known to send 183s with no SDP after sending
+@@ -20371,6 +20595,10 @@
/* Check for Session-Timers related headers */
if (st_get_mode(p, 0) != SESSION_TIMER_MODE_REFUSE && p->outgoing_call == TRUE && !reinvite) {
@@ -451,16 +537,17 @@
p_hdrval = (char*)get_header(req, "Session-Expires");
if (!ast_strlen_zero(p_hdrval)) {
/* UAS supports Session-Timers */
-@@ -21068,6 +21252,8 @@
+@@ -21073,6 +21301,9 @@
struct ast_channel *owner;
int sipmethod;
const char *c = get_header(req, "Cseq");
+ const char *required = get_header(req, "Require");
++ char tag[128];
+
/* GCC 4.2 complains if I try to cast c as a char * when passing it to ast_skip_nonblanks, so make a copy of it */
char *c_copy = ast_strdupa(c);
/* Skip the Cseq and its subsequent spaces */
-@@ -21109,7 +21295,7 @@
+@@ -21114,7 +21345,7 @@
ack_res = __sip_semi_ack(p, seqno, 0, sipmethod);
}
} else {
@@ -469,15 +556,28 @@
}
if (ack_res == FALSE) {
-@@ -21135,6 +21321,7 @@
- gettag(req, "To", tag, sizeof(tag));
- ast_string_field_set(p, theirtag, tag);
- }
-+
+@@ -21133,13 +21364,14 @@
+ p->pendinginvite = 0;
+ }
+
+- /* Get their tag if we haven't already */
+- if (ast_strlen_zero(p->theirtag) || (resp >= 200)) {
+- char tag[128];
++
++ /* Always get the tag. Find_call will filter out after we have an established dialog,
++ so that we don't update the tag after a 200 or other final response.
++ Provided that SIP pedantic checking is turned on of course.
++ */
++ gettag(req, "To", tag, sizeof(tag));
++ ast_string_field_set(p, theirtag, tag);
+
+- gettag(req, "To", tag, sizeof(tag));
+- ast_string_field_set(p, theirtag, tag);
+- }
/* 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.
-@@ -21154,7 +21341,44 @@
+@@ -21159,7 +21391,50 @@
pvt_set_needdestroy(p, "received 4XX response to a BYE");
return;
}
@@ -519,10 +619,16 @@
+ }
+ }
+
++ if (sip_cfg.early_media_focus && !ast_strlen_zero(p->theirtag_early) && strcmp(p->theirtag_early, p->theirtag)) {
++ /* If we already are in early media phase, and have a response from a new device in this call we should
++ ignore the SDP. */
++ req->ignoresdp = TRUE;
++ }
++
if (p->relatedpeer && sipmethod == SIP_OPTIONS) {
/* We don't really care what the response is, just that it replied back.
Well, as long as it's not a 100 response... since we might
-@@ -21171,6 +21395,9 @@
+@@ -21176,6 +21451,9 @@
} else if (sipmethod == SIP_INFO) {
/* More good gravy! */
handle_response_info(p, resp, rest, req, seqno);
@@ -532,7 +638,7 @@
} else if (sipmethod == SIP_MESSAGE) {
/* More good gravy! */
handle_response_message(p, resp, rest, req, seqno);
-@@ -22318,6 +22545,37 @@
+@@ -22323,6 +22601,37 @@
return 0;
}
@@ -570,7 +676,7 @@
/*!
* \brief Handle incoming INVITE request
* \note If the INVITE has a Replaces header, it is part of an
-@@ -22379,7 +22637,7 @@
+@@ -22384,7 +22693,7 @@
p->invitestate = INV_COMPLETED;
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -579,7 +685,7 @@
goto request_invite_cleanup;
}
}
-@@ -22389,6 +22647,22 @@
+@@ -22394,6 +22703,23 @@
p->sipoptions |= required_profile;
p->reqsipoptions = required_profile;
@@ -587,6 +693,7 @@
+ if (p->reqsipoptions & SIP_OPT_100REL || p->sipoptions & SIP_OPT_100REL) {
+ if (ast_test_flag(&p->flags[2], SIP_PAGE3_PRACK)) { /* Is PRACK enabled for this dialog? */
+ ast_set_flag(&p->flags[2], SIP_PAGE3_100REL); /* Mark PRACK as active for this dialog */
++ ast_debug(2, "--#-#-#-#- Adding PRACK support for this dialog \n");
+ } else if (p->reqsipoptions & SIP_OPT_100REL) {
+ /* If PRACK was required but is disabled in configuration, don't play */
+ transmit_response(p, "420 Bad extension (unsupported)", req);
@@ -602,7 +709,7 @@
/* Check if this is a loop */
if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && p->owner && (p->invitestate != INV_TERMINATED && p->invitestate != INV_CONFIRMED) && p->owner->_state != AST_STATE_UP) {
/* This is a call to ourself. Send ourselves an error code and stop
-@@ -22451,7 +22725,7 @@
+@@ -22456,7 +22782,7 @@
* transaction. Calling __sip_ack will take care of this by clearing the p->pendinginvite and removing the response
* from the previous transaction from the list of outstanding packets.
*/
@@ -611,7 +718,7 @@
} else {
/* We already have a pending invite. Sorry. You are on hold. */
p->glareinvite = seqno;
-@@ -24547,7 +24821,7 @@
+@@ -24552,7 +24878,7 @@
return 0;
} else if (auth_result == AUTH_SUCCESSFUL && p->lastinvite) {
/* We need to stop retransmitting the 401 */
@@ -620,7 +727,7 @@
}
publish_type = determine_sip_publish_type(req, event, etag, expires_str, &expires_int);
-@@ -25374,12 +25648,15 @@
+@@ -25379,12 +25705,15 @@
case SIP_UPDATE:
res = handle_request_update(p, req);
break;
@@ -637,7 +744,7 @@
if (find_sdp(req)) {
if (process_sdp(p, req, SDP_T38_NONE))
return -1;
-@@ -25388,7 +25665,7 @@
+@@ -25393,7 +25722,7 @@
} else if (p->glareinvite == seqno) {
/* handle ack for the 491 pending sent for glareinvite */
p->glareinvite = 0;
@@ -646,7 +753,7 @@
}
if (!acked) {
/* Got an ACK that did not match anything. Ignore
-@@ -27032,6 +27309,9 @@
+@@ -27037,6 +27366,9 @@
} else if (!strcasecmp(v->name, "buggymwi")) {
ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
@@ -656,10 +763,27 @@
} else
res = 0;
+@@ -28244,6 +28576,7 @@
+ externtcpport = STANDARD_SIP_PORT;
+ externtlsport = STANDARD_TLS_PORT;
+ sip_cfg.srvlookup = DEFAULT_SRVLOOKUP;
++ sip_cfg.early_media_focus = DEFAULT_EARLY_MEDIA_FOCUS;
+ global_tos_sip = DEFAULT_TOS_SIP;
+ global_tos_audio = DEFAULT_TOS_AUDIO;
+ global_tos_video = DEFAULT_TOS_VIDEO;
+@@ -28597,6 +28930,8 @@
+ global_match_auth_username = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "srvlookup")) {
+ sip_cfg.srvlookup = ast_true(v->value);
++ } else if (!strcasecmp(v->name, "earlymediafocus")) {
++ sip_cfg.early_media_focus = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "pedantic")) {
+ sip_cfg.pedanticsipchecking = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "maxexpirey") || !strcasecmp(v->name, "maxexpiry")) {
Index: channels/sip/include/sip.h
===================================================================
---- channels/sip/include/sip.h (.../branches/1.8) (revision 369624)
-+++ channels/sip/include/sip.h (.../team/oej/darjeeling-prack-1.8) (revision 369624)
+--- channels/sip/include/sip.h (.../branches/1.8) (revision 369789)
++++ channels/sip/include/sip.h (.../team/oej/darjeeling-prack-1.8) (revision 369789)
@@ -155,7 +155,7 @@
* \todo This string should be set dynamically. We only support REFER and SUBSCRIBE if we have
* allowsubscribe and allowrefer on in sip.conf.
@@ -669,7 +793,15 @@
/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_SIP_PORT 5060
-@@ -223,6 +223,7 @@
+@@ -187,6 +187,7 @@
+ #define DEFAULT_MWI_FROM ""
+ #define DEFAULT_NOTIFYMIME "application/simple-message-summary"
+ #define DEFAULT_ALLOWGUEST TRUE
++#define DEFAULT_EARLY_MEDIA_FOCUS FALSE; /*!< Focus on a single early media stream */
+ #define DEFAULT_RTPKEEPALIVE 0 /*!< Default RTPkeepalive setting */
+ #define DEFAULT_CALLCOUNTER FALSE /*!< Do not enable call counters by default */
+ #define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */
+@@ -223,6 +224,7 @@
#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_TESTLAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
#define DEFAULT_STORE_SIP_CAUSE FALSE /*!< Don't store HASH(SIP_CAUSE,<channel name>) for channels by default */
#endif
@@ -677,7 +809,7 @@
/*@}*/
/*! \name SIPflags
-@@ -356,10 +357,14 @@
+@@ -356,10 +358,14 @@
SIP_PAGE2_Q850_REASON | SIP_PAGE2_HAVEPEERCONTEXT | SIP_PAGE2_USE_SRTP)
@@ -694,7 +826,7 @@
/*@}*/
-@@ -403,6 +408,7 @@
+@@ -403,6 +409,7 @@
* where the original response would be sent RELIABLE in an INVITE transaction
*/
enum xmittype {
@@ -702,7 +834,7 @@
XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
* If it fails, it's critical and will cause a teardown of the session */
XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */
-@@ -560,7 +566,7 @@
+@@ -560,7 +567,7 @@
SIP_NOTIFY, /*!< Status update, Part of the event package standard, result of a SUBSCRIBE or a REFER */
SIP_INVITE, /*!< Set up a session */
SIP_ACK, /*!< End of a three-way handshake started with INVITE. */
@@ -711,7 +843,15 @@
SIP_BYE, /*!< End of a session */
SIP_REFER, /*!< Refer to another URI (transfer) */
SIP_SUBSCRIBE, /*!< Subscribe for updates (voicemail, session status, device status, presence) */
-@@ -749,6 +755,8 @@
+@@ -675,6 +682,7 @@
+ be applied to devices (trunks, services, phones)
+ */
+ struct sip_settings {
++ int early_media_focus; /*!< G: Focus on the first early media stream received, ignore the rest */
+ int peer_rtupdate; /*!< G: Update database with registration data for peer? */
+ int rtsave_sysname; /*!< G: Save system name at registration? */
+ int ignore_regexpire; /*!< G: Ignore expiration of peer */
+@@ -749,12 +757,15 @@
int headers; /*!< # of SIP Headers */
int method; /*!< Method of this request */
int lines; /*!< Body Content */
@@ -720,7 +860,23 @@
unsigned int sdp_start; /*!< the line number where the SDP begins */
unsigned int sdp_count; /*!< the number of lines of SDP */
char debug; /*!< print extra debugging if non zero */
-@@ -985,6 +993,8 @@
+ char has_to_tag; /*!< non-zero if packet has To: tag */
+ char ignore; /*!< if non-zero This is a re-transmit, ignore it */
+ char authenticated; /*!< non-zero if this request was authenticated */
++ char ignoresdp; /*!< In some cases, we have to ignore the SDP in responses */
+ ptrdiff_t header[SIP_MAX_HEADERS]; /*!< Array of offsets into the request string of each SIP header*/
+ ptrdiff_t line[SIP_MAX_LINES]; /*!< Array of offsets into the request string of each SDP line*/
+ struct ast_str *data;
+@@ -959,6 +970,8 @@
+ AST_STRING_FIELD(rdnis); /*!< Referring DNIS */
+ AST_STRING_FIELD(redircause); /*!< Referring cause */
+ AST_STRING_FIELD(theirtag); /*!< Their tag */
++ AST_STRING_FIELD(theirtag_prack); /*!< Current tag focus for PRACK handling */
++ AST_STRING_FIELD(theirtag_early); /*!< Current tag focus for early media handling */
+ 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] */
+@@ -985,6 +998,8 @@
uint32_t ocseq; /*!< Current outgoing seqno */
uint32_t icseq; /*!< Current incoming seqno */
uint32_t init_icseq; /*!< Initial incoming seqno from first request */
@@ -729,7 +885,7 @@
ast_group_t callgroup; /*!< Call group */
ast_group_t pickupgroup; /*!< Pickup group */
uint32_t lastinvite; /*!< Last seqno of invite */
-@@ -1142,6 +1152,7 @@
+@@ -1142,6 +1157,7 @@
int retrans; /*!< Retransmission number */
int method; /*!< SIP method for this packet */
uint32_t seqno; /*!< Sequence number */
@@ -737,7 +893,7 @@
char is_resp; /*!< 1 if this is a response packet (e.g. 200 OK), 0 if it is a request */
char is_fatal; /*!< non-zero if there is a fatal error */
int response_code; /*!< If this is a response, the response code */
-@@ -1751,7 +1762,7 @@
+@@ -1751,7 +1767,7 @@
char * const text; /*!< Text id, as in standard */
} sip_options[] = { /* XXX used in 3 places */
/* RFC3262: PRACK 100% reliability */
@@ -748,8 +904,8 @@
/* SIMPLE events: RFC4662 */
Index: channels/sip/include/reqresp_parser.h
===================================================================
---- channels/sip/include/reqresp_parser.h (.../branches/1.8) (revision 369624)
-+++ channels/sip/include/reqresp_parser.h (.../team/oej/darjeeling-prack-1.8) (revision 369624)
+--- channels/sip/include/reqresp_parser.h (.../branches/1.8) (revision 369789)
++++ channels/sip/include/reqresp_parser.h (.../team/oej/darjeeling-prack-1.8) (revision 369789)
@@ -145,6 +145,14 @@
unsigned int parse_sip_options(const char *options, char *unsupported, size_t unsupported_len);
@@ -767,8 +923,8 @@
* \param input1 First URI
Index: channels/sip/include/dialog.h
===================================================================
---- channels/sip/include/dialog.h (.../branches/1.8) (revision 369624)
-+++ channels/sip/include/dialog.h (.../team/oej/darjeeling-prack-1.8) (revision 369624)
+--- channels/sip/include/dialog.h (.../branches/1.8) (revision 369789)
++++ channels/sip/include/dialog.h (.../team/oej/darjeeling-prack-1.8) (revision 369789)
@@ -67,7 +67,7 @@
/*! \brief Acknowledges receipt of a packet and stops retransmission
@@ -780,8 +936,8 @@
* called with p locked */
Index: channels/sip/reqresp_parser.c
===================================================================
---- channels/sip/reqresp_parser.c (.../branches/1.8) (revision 369624)
-+++ channels/sip/reqresp_parser.c (.../team/oej/darjeeling-prack-1.8) (revision 369624)
+--- channels/sip/reqresp_parser.c (.../branches/1.8) (revision 369789)
++++ channels/sip/reqresp_parser.c (.../team/oej/darjeeling-prack-1.8) (revision 369789)
@@ -1516,18 +1516,13 @@
}
@@ -852,7 +1008,7 @@
Index: README.darjeeling
===================================================================
--- README.darjeeling (.../branches/1.8) (revision 0)
-+++ README.darjeeling (.../team/oej/darjeeling-prack-1.8) (revision 369624)
++++ README.darjeeling (.../team/oej/darjeeling-prack-1.8) (revision 369789)
@@ -0,0 +1,130 @@
+Edvina AB
+Olle E. Johansson
@@ -996,25 +1152,40 @@
Index: configs/sip.conf.sample
===================================================================
---- configs/sip.conf.sample (.../branches/1.8) (revision 369624)
-+++ configs/sip.conf.sample (.../team/oej/darjeeling-prack-1.8) (revision 369624)
+--- configs/sip.conf.sample (.../branches/1.8) (revision 369789)
++++ configs/sip.conf.sample (.../team/oej/darjeeling-prack-1.8) (revision 369789)
@@ -241,6 +241,9 @@
; and multiline formatted headers for strict
; SIP compatibility (defaults to "yes")
-+;prack=yes ; Enable PRACK support. (defaults to "no")
++;prack=yes ; Enable PRACK (SIP option 100rel) support. (defaults to "no")
+ ; Can also be set on a per device basis.
+
; See https://wiki.asterisk.org/wiki/display/AST/IP+Quality+of+Service for a description of these parameters.
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
+@@ -346,6 +349,14 @@
+ ;
+ ; In order for "noanswer" applications to work, you need to run
+ ; the progress() application in the priority before the app.
++;earlymediafocus=yes ; If you have a forking SIP proxy in the call, Asterisk may get multiple
++ ; 183/180 responses with SDP from *different* devices. The default is that
++ ; Asterisk jumps on to each one. There's no *correct* solution to this
++ ; problem. Turning this on means that Asterisk focuses on the *first*
++ ; early media response and ignores the rest. This might mean that the
++ ; user gets a ring tone and stays with that even if the service provider
++ ; from a different media server plays a very important message.
++ ; Default is "no" - jump to the latest one.
+
+ ;progressinband=never ; If we should generate in-band ringing always
+ ; use 'never' to never use in-band signalling, even in cases
Egenskapsändringar för: .
___________________________________________________________________
Added: automerge
+ *
Added: svnmerge-integrated
- + /branches/1.8:1-369590
+ + /branches/1.8:1-369757
Added: automerge-email
+ oej at edvina.net
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