[svn-commits] oej: branch oej/darjeeling-prack-11 r383212 - in /team/oej/darjeeling-prack-1...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Mar 15 09:07:06 CDT 2013
Author: oej
Date: Fri Mar 15 09:07:02 2013
New Revision: 383212
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=383212
Log:
Upgrade PRACK work for Asterisk 11
Added:
team/oej/darjeeling-prack-11/README.darjeeling (with props)
Modified:
team/oej/darjeeling-prack-11/channels/chan_sip.c
team/oej/darjeeling-prack-11/channels/sip/include/dialog.h
team/oej/darjeeling-prack-11/channels/sip/include/reqresp_parser.h
team/oej/darjeeling-prack-11/channels/sip/include/sip.h
team/oej/darjeeling-prack-11/channels/sip/reqresp_parser.c
team/oej/darjeeling-prack-11/configs/sip.conf.sample
Added: team/oej/darjeeling-prack-11/README.darjeeling
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-11/README.darjeeling?view=auto&rev=383212
==============================================================================
--- team/oej/darjeeling-prack-11/README.darjeeling (added)
+++ team/oej/darjeeling-prack-11/README.darjeeling Fri Mar 15 09:07:02 2013
@@ -1,0 +1,129 @@
+Edvina AB
+Olle E. Johansson
+
+
+Project started: 2012-06-15
+Updated to Asterisk 11: 2013-03-15
+
+
+
+Darjeeling-prack-11
+-------------------
+
+This branch will implement PRACK in the SIP stack of Asterisk.
+
+PRACK stands for reliable unreliable provisional responses.
+
+In SIP, the provisional responses are often sent over UDP, which means that they can
+get lost. Some of them are retransmitted every minute (to get at least one through
+once during a three minute period following RFC 3261). For some messages, it's
+important that they get through immediately. Like if you want to play a message to
+the customer that his call will be cancelled due to lack of funds in his account.
+
+PRACK adds a retransmit and ACK mechanism to the 1xx messages excluding 100 (since
+it's transmitted hop-by-hop).
+
+Configuration
+=============
+Add prack=yes in the [general] section of sip.conf or in device configurations.
+Asterisk will now add 100rel to the list of supported options. If the other device
+supports PRACK Asterisk will activate it or support it if the other side
+requires it.
+There's currently no support for requiring PRACK in a call.
+
+
+Technichal details
+==================
+
+If the INVITE contains
+ Supported: 100rel
+
+then the 1xx answer can add
+ Require: 100rel
+ Rseq: 42
+
+The Rseq is the PRACK sequence number
+The caller then needs to confirm the message with a new request, during the INVITE transaction
+ PRACK
+ RAck: 42 <cseq #> <cseq method>
+
+And the callee confirms this (and close the PRACK transaction) with a 200 OK.
+
+If the PRACK is not received in time, the 1xx response will be retransmitted.
+There can only be ONE outstanding PRACK, which makes it easier to integrate in the
+Asterisk SIP stack that unfortunately lacks a transaction layer.
+
+PRACK is documented in RFC 3262.
+
+Retransmission works like the retransmission of responses to INVITE (like the 200 OK).
+It starts with T1 and doubles for each retransmission. Unlike INVITE responses,
+the retransmission timer does not cap at T2.
+
+If retransmission times out, a 5xx message should terminate the INVITE transaction.
+
+
+A PRACK received that does not match existing SIP_PVT is responded to with 481.
+
+* When to stop?
+---------------
+From the RFC:
+"The UAS MAY send a final response to the initial request before having received PRACKs for all unacknowledged
+reliable provisional responses, unless the final response is 2xx and any of the unacknowledged
+reliable provisional responses contained a session description. In that case, it MUST NOT send a final response
+until those provisional responses are acknowledged. If the UAS does send a final response when
+reliable responses are still unacknowledged, it SHOULD NOT continue to retransmit the unacknowledged
+reliable provisional responses, but it MUST be prepared to process PRACK requests for those outstanding
+responses. A UAS MUST NOT send new reliable provisional responses (as opposed to retransmissions of
+unacknowledged ones) after sending a final response to a request"
+
+* Receive in order until final response
+---------------------------------------
+"Handling of subsequent reliable provisional responses for the same initial request follows the same rules
+as above, with the following difference: reliable provisional responses are guaranteed to be in order. As a
+result, 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.
+The UAC MAY acknowledge reliable provisional responses received after the final response or MAY discard
+them."
+
+* When do the call begin?
+-------------------------
+A corner case that I don't know if it's implemented: If we receive and INVITE without SDP, we MUST
+add SDP offer to the reliable answer and the caller must add an SDP answer to the PRACK. This means
+that the session begins in the middle of the INVITE transaction.
+"Once an answer has been sent or received, the UA SHOULD establish the session based on the parameters of
+the offer and answer, even if the original INVITE itself has not been responded to." (section 5)
+
+* Security
+----------
+"The PRACK request can be injected by attackers to force retransmissions of reliable provisional responses to
+cease. As these responses can convey important information, PRACK messages SHOULD be authenticated
+as any other request." (section 9)
+
+Todo
+----
+* Add PRACK to the list of supported headers
+ - done
+* Should we be able to REQUIRE prack? Based on what? _SIPREQUIREPRACK ?
+ - not now
+* PRACK enabled globally and per device - user and peer
+ - done
+* PRACK working when Asterisk is the UAS
+ - done
+* PRACK working when Asterisk is the UAC
+ - done
+* PRACK with authentication
+ - not done
+* Requesting auth for PRACK
+ - not done
+* PRACK in "sip show settings"
+ - done
+* PRACK in "sip show peer xxx"
+ - done
+* PRACK in "sip show channel xxx"
+ - done
+
+---------------------
+The patch is (C) Copyright by Edvina AB, Sollentuna, Sweden.
+Developed by Olle E. Johansson, oej at edvina.net
Propchange: team/oej/darjeeling-prack-11/README.darjeeling
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/oej/darjeeling-prack-11/README.darjeeling
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/oej/darjeeling-prack-11/README.darjeeling
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: team/oej/darjeeling-prack-11/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-11/channels/chan_sip.c?view=diff&rev=383212&r1=383211&r2=383212
==============================================================================
--- team/oej/darjeeling-prack-11/channels/chan_sip.c (original)
+++ team/oej/darjeeling-prack-11/channels/chan_sip.c Fri Mar 15 09:07:02 2013
@@ -1325,10 +1325,10 @@
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);
static int transmit_response_using_temp(ast_string_field callid, struct ast_sockaddr *addr, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg);
-static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp, int rpid);
+static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale);
static int transmit_provisional_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, int with_sdp);
@@ -1349,6 +1349,7 @@
static int transmit_cc_notify(struct ast_cc_agent *agent, struct sip_pvt *subscription, enum sip_cc_notify_state state);
static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno);
+static void add_required_respheader(struct sip_request *req);
static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno);
static void copy_request(struct sip_request *dst, const struct sip_request *src);
static void receive_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
@@ -4308,6 +4309,9 @@
if (sscanf(ast_str_buffer(pkt->data), "SIP/2.0 %30u", &respid) == 1) {
pkt->response_code = respid;
}
+ if (ast_test_flag(&p->flags[2], SIP_PAGE3_100REL) && respid > 100 && respid < 200) {
+ pkt->rseqno = p->rseq;
+ }
}
pkt->timer_t1 = p->timer_t1; /* Set SIP timer T1 */
pkt->retransid = -1;
@@ -4492,7 +4496,7 @@
/*! \brief Acknowledges receipt of a packet and stops retransmission
* called with p locked*/
-int __sip_ack(struct sip_pvt *p, uint32_t seqno, int resp, int sipmethod)
+int __sip_ack(struct sip_pvt *p, uint32_t seqno, int resp, int sipmethod, uint32_t rseqno)
{
struct sip_pkt *cur, *prev = NULL;
const char *msg = "Not Found"; /* used only for debugging */
@@ -4511,6 +4515,10 @@
if (cur->seqno != seqno || cur->is_resp != resp) {
continue;
}
+ /* With PRACK we can have a situation with multiple unPRACKed responses */
+ if (rseqno && cur->rseqno != rseqno) {
+ continue;
+ }
if (cur->is_resp || cur->method == sipmethod) {
res = TRUE;
msg = "Found";
@@ -4522,6 +4530,7 @@
if (sipdebug)
ast_debug(4, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
}
+
/* This odd section is designed to thwart a
* race condition in the packet scheduler. There are
* two conditions under which deleting the packet from the
@@ -4552,8 +4561,8 @@
break;
}
}
- 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 %u\n",
+ p->callid, resp ? "Response" : "Request", seqno, msg, rseqno);
return res;
}
@@ -4571,7 +4580,7 @@
}
cur = p->packets;
method = (cur->method) ? cur->method : find_sip_method(ast_str_buffer(cur->data));
- __sip_ack(p, cur->seqno, cur->is_resp, method);
+ __sip_ack(p, cur->seqno, cur->is_resp, method, cur->rseqno);
}
}
@@ -4686,6 +4695,10 @@
with_sdp ? send_provisional_keepalive_with_sdp : send_provisional_keepalive, dialog_ref(pvt, "Increment refcount to pass dialog pointer to sched callback"));
}
+/*! \brief Adds a Required header
+
+ Needs to be called before attachment (i.e. SDP) is added
+*/
static void add_required_respheader(struct sip_request *req)
{
struct ast_str *str;
@@ -4714,10 +4727,34 @@
ast_free(str);
}
+/*! \brief Active PRACK if supported by config and by other end */
+static void add_prack_respheader(struct sip_pvt *p, struct sip_request *req, int reliable)
+{
+ /* If method is INVITE and it contains Supported: 100 rel and we have enabled PRACK */
+ if (ast_test_flag(&p->flags[2], SIP_PAGE3_100REL)) {
+ /* Check if the invite has 100 REL supported here */
+ if (reliable == XMIT_PRACK) {
+ char buf[SIPBUFSIZE/2];
+ if (p->rseq == 0) {
+ p->rseq = 41; /* Starting level. Hi Douglas */
+ }
+ snprintf(buf, sizeof(buf), "%d", ++(p->rseq));
+ add_header(req, "Rseq", buf);
+ req->rseqno = p->rseq;
+ req->reqsipoptions |= SIP_OPT_100REL;
+ append_history(p, "PRACK", "PRACK Required: Our Rseq %u", p->rseq);
+ ast_debug(2, "=!=!=!=!=!=!=!= PRACK USED HERE. Rseq %u \n", p->rseq);
+ } else {
+ ast_debug(2, "=!=!=!=!=!=!=!= PRACK COULD BE USED HERE. Exactly HERE\n");
+ }
+ }
+}
+
/*! \brief Transmit response on SIP request*/
static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno)
{
int res;
+
finalize_content(req);
add_blank(req);
@@ -4743,7 +4780,7 @@
}
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) {
@@ -7249,9 +7286,14 @@
struct sip_pvt *p = ast_channel_tech_pvt(ast);
sip_pvt_lock(p);
- if (ast_channel_state(ast) != AST_STATE_UP) {
+ if ((ast_channel_state(ast) != AST_STATE_UP) && ast_test_flag(&p->flags[2], SIP_PAGE3_INVITE_WAIT_FOR_PRACK)) {
+ ast_set_flag(&p->flags[2], SIP_PAGE3_ANSWER_WAIT_FOR_PRACK);
+ ast_debug(2, "<-<-<-<-<-<-< HOLDING Answer while waiting for PRACK to arrive on channel %s\n", ast_channel_name(ast));
+ sip_pvt_unlock(p);
+ return 0;
+ }
+ if ((ast_channel_state(ast) != AST_STATE_UP) || ast_test_flag(&p->flags[2], SIP_PAGE3_INVITE_WAIT_FOR_PRACK)) {
try_suggested_sip_codec(p);
-
ast_setstate(ast, AST_STATE_UP);
ast_debug(1, "SIP answering channel: %s\n", ast_channel_name(ast));
ast_rtp_instance_update_source(p->rtp);
@@ -11283,13 +11325,15 @@
* is supported for this dialog. */
static int add_supported(struct sip_pvt *pvt, struct sip_request *req)
{
- int res;
+ char supported[256] = "replaces";
+
if (st_get_mode(pvt, 0) != SESSION_TIMER_MODE_REFUSE) {
- res = add_header(req, "Supported", "replaces, timer");
- } else {
- res = add_header(req, "Supported", "replaces");
- }
- return res;
+ strncat(supported, ", timer", sizeof(supported));
+ }
+ if (ast_test_flag(&pvt->flags[2], SIP_PAGE3_PRACK)) {
+ strncat(supported, ", 100rel", sizeof(supported));
+ }
+ return add_header(req, "Supported", supported);
}
/*! \brief Add header to SIP message */
@@ -11930,7 +11974,9 @@
struct sip_request resp;
uint32_t seqno = 0;
- if (reliable && (sscanf(sip_get_header(req, "CSeq"), "%30u ", &seqno) != 1)) {
+ int res = sscanf(sip_get_header(req, "CSeq"), "%30u ", &seqno);
+
+ if (reliable && res != 1) {
ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", sip_get_header(req, "CSeq"));
return -1;
}
@@ -11941,6 +11987,10 @@
&& (!strncmp(msg, "180", 3) || !strncmp(msg, "183", 3))) {
ast_clear_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
add_rpid(&resp, p);
+ }
+ if (ast_test_flag(&p->flags[2], SIP_PAGE3_100REL) && strncmp(msg, "100", 3) && !strncmp(msg, "1", 1)) {
+ ast_debug(2, "=!=!=!=!=!= PRACK applied to message \"%s\" \n", msg);
+ reliable = XMIT_PRACK;
}
if (ast_test_flag(&p->flags[0], SIP_OFFER_CC)) {
add_cc_call_info_to_response(p, &resp);
@@ -11981,6 +12031,10 @@
add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
}
}
+ if (strncmp(msg, "100", 3)) {
+ add_prack_respheader(p, &resp, reliable);
+ add_required_respheader(&resp);
+ }
return send_response(p, &resp, reliable, seqno);
}
@@ -11993,6 +12047,7 @@
}
respprep(&resp, p, msg, req);
add_header(&resp, "SIP-ETag", esc_entry->entity_tag);
+ add_required_respheader(&resp);
return send_response(p, &resp, 0, 0);
}
@@ -12081,6 +12136,7 @@
respprep(&resp, p, msg, req);
add_date(&resp);
add_header(&resp, "Unsupported", unsupported);
+ add_required_respheader(&resp);
return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
@@ -12152,6 +12208,7 @@
struct sip_request resp;
respprep(&resp, p, msg, req);
add_header(&resp, "Accept", "application/sdp");
+ add_required_respheader(&resp);
return send_response(p, &resp, reliable, 0);
}
@@ -12164,6 +12221,7 @@
snprintf(tmp, sizeof(tmp), "%d", minexpires);
respprep(&resp, p, msg, req);
add_header(&resp, "Min-Expires", tmp);
+ add_required_respheader(&resp);
return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
@@ -13521,6 +13579,18 @@
if (rpid == TRUE) {
add_rpid(&resp, p);
}
+ if (ast_test_flag(&p->flags[2], SIP_PAGE3_100REL) && strncmp(msg, "100", 3) && !strncmp(msg, "1", 1)) {
+ ast_debug(2, "=!=!=!=!=!= PRACK applied to message \"%s\" \n", msg);
+ reliable = XMIT_PRACK;
+ }
+ if (strncmp(msg, "100", 3)) {
+ /* 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);
}
@@ -13959,7 +14029,69 @@
}
/*!
- * \brief Build REFER/INVITE/OPTIONS/SUBSCRIBE message and transmit it
+ * \brief transmit SIP PRACK as a response to a provisional response with a Rseq and Require: 100rel header
+ */
+static int transmit_prack(struct sip_pvt *p, uint32_t their_rseq)
+{
+ 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.
+ */
+ 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;
+ 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;
+}
+
+/*!
+ * \brief Build PRACK/REFER/INVITE/OPTIONS/SUBSCRIBE message and transmit it
* \param p sip_pvt structure
* \param sipmethod
* \param sdp unknown
@@ -13974,7 +14106,9 @@
if (init) {/* Bump branch even on initial requests */
p->branch ^= ast_random();
- p->invite_branch = p->branch;
+ if (sipmethod != SIP_PRACK) {
+ p->invite_branch = p->branch;
+ }
build_via(p);
}
if (init > 1) {
@@ -14004,12 +14138,18 @@
add_header(&req, "Accept", "application/call-completion");
}
add_expires(&req, p->expiry);
+ } else if (sipmethod == SIP_PRACK) {
+ /* Add headers for PRACK */
+ char buf[SIPBUFSIZE/2];
+ snprintf(buf, sizeof(buf), "%u %u %s", p->irseq, p->lastinvite, "INVITE");
+ add_header(&req, "RAck", buf);
}
/* This new INVITE is part of an attended transfer. Make sure that the
other end knows and replace the current call with this new call */
if (p->options && !ast_strlen_zero(p->options->replaces)) {
add_header(&req, "Replaces", p->options->replaces);
+ /* XXX This needs to be automated since we can have multiple options here */
add_header(&req, "Require", "replaces");
}
@@ -19830,6 +19970,7 @@
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)));
+ ast_cli(fd, " PRACK support: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[2], SIP_PAGE3_PRACK)));
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)));
@@ -19966,6 +20107,7 @@
/* - is enumerated */
astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
+ astman_append(s, "SIP-PRACK: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_PRACK) ? "Y" : "N");
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));
@@ -20567,7 +20709,9 @@
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");
@@ -20951,6 +21095,10 @@
ast_cli(a->fd, " MaxCallBR: %d kbps\n", cur->maxcallbitrate);
ast_cli(a->fd, " Theoretical Address: %s\n", ast_sockaddr_stringify(&cur->sa));
ast_cli(a->fd, " Received Address: %s\n", ast_sockaddr_stringify(&cur->recv));
+ ast_cli(a->fd, " SIP PRACK support: %s\n", ast_test_flag(&cur->flags[2], SIP_PAGE3_100REL) ? "Active" :
+ (ast_test_flag(&cur->flags[2], SIP_PAGE3_PRACK) ? "Enabled" : "Disabled"));
+
+ ast_cli(a->fd, " SIP PRACK active %s\n", AST_CLI_YESNO(ast_test_flag(&cur->flags[2], SIP_PAGE3_100REL)));
ast_cli(a->fd, " SIP Transfer mode: %s\n", transfermode2str(cur->allowtransfer));
ast_cli(a->fd, " Force rport: %s\n", force_rport_string(cur->flags));
if (ast_sockaddr_isnull(&cur->redirip)) {
@@ -22266,6 +22414,48 @@
return 0;
}
+/*! \brief Handle PRACK responses
+ */
+static void handle_response_prack(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
+{
+ ast_debug(2, "---> Got response on PRACK :: %d \n", resp);
+ /* Handle authentication early */
+ if (resp == 401 || resp == 407) {
+ if (p->options) {
+ p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
+ }
+ if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp, SIP_PRACK, 1)) {
+ ast_log(LOG_NOTICE, "Failed to authenticate on PRACK to '%s'\n", sip_get_header(&p->initreq, "From"));
+ }
+ return;
+ }
+
+ /* THe REALLY important thing is that the PRACK request gets a response. The response itself
+ is not that important. A 481 means that the call will hang up. No response at all means
+ that the call will hang up
+ */
+ switch(resp) {
+ case 200: /* 200 OK - all is fine in the kingdom of SIP */
+ break;
+
+ case 408: /* Timeout */
+ case 481: /* Ok, they did not find our call ID. Let's die */
+ if (p->owner) {
+ ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
+ }
+ break;
+ case 403: /* Forbidden */
+ case 415: /* Unsupported media type */
+ case 488: /* Not acceptable here */
+ case 606: /* Not Acceptable */
+ default:
+ /* Don't do anything */
+ break;
+ };
+
+
+}
+
/*!
* \brief Handle authentication challenge for SIP UPDATE
*
@@ -22535,7 +22725,7 @@
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;
}
@@ -22545,6 +22735,9 @@
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;
@@ -22618,7 +22811,7 @@
}
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;
}
@@ -22626,6 +22819,9 @@
if (!req->ignore && p->owner) {
/* 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 {
@@ -23512,6 +23708,8 @@
char *c_copy = ast_strdupa(c);
/* Skip the Cseq and its subsequent spaces */
const char *msg = ast_skip_blanks(ast_skip_nonblanks(c_copy));
+ const char *required = sip_get_header(req, "Require");
+ char tag[128];
if (!msg)
msg = "";
@@ -23550,7 +23748,7 @@
ack_res = __sip_semi_ack(p, seqno, 0, sipmethod);
}
} else {
- ack_res = __sip_ack(p, seqno, 0, sipmethod);
+ ack_res = __sip_ack(p, seqno, 0, sipmethod, 0);
}
if (ack_res == FALSE) {
@@ -23569,13 +23767,14 @@
p->pendinginvite = 0;
}
- /* Get their tag if we haven't already */
- if (ast_strlen_zero(p->theirtag) || (resp >= 200)) {
- char tag[128];
-
- gettag(req, "To", tag, sizeof(tag));
- ast_string_field_set(p, theirtag, tag);
- }
+
+ /* 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);
+
/* 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.
@@ -23594,6 +23793,50 @@
if ((resp == 404 || resp == 408 || resp == 481) && sipmethod == SIP_BYE) {
pvt_set_needdestroy(p, "received 4XX response to a BYE");
return;
+ }
+
+ /* If we have a required header in the response, the other side have activated an extension
+ we said that we do support */
+ if (!ast_strlen_zero(required)) {
+ int activeextensions = parse_required_sip_options(required);
+ if (activeextensions & SIP_OPT_100REL) {
+
+ const char *rseq = sip_get_header(req, "RSeq");
+ uint32_t their_rseq;
+ int res;
+ ast_debug(3, "!=!=!=!=!=! Response relies on PRACK! Rseq %s\n", rseq);
+
+ /* XXX If the response relies on PRACK, we need to start a PRACK transaction
+ */
+ sscanf(sip_get_header(req, "RSeq"), "%30u ", &their_rseq);
+ append_history(p, "TxPrack", "Their Rseq %u\n", their_rseq);
+ parse_ok_contact(p, req);
+ build_route(p, req, 1, resp);
+
+ res = transmit_prack(p, their_rseq);
+ if (res == -2) {
+ /* This response is a retransmit and should be ignored */
+ /* RFC 3262: Once a reliable provisional response is received, retransmissions of
+ that response MUST be discarded. A response is a retransmission when
+ its dialog ID, CSeq, and RSeq match the original response.
+ */
+ append_history(p, "PrIgnore", "Ignoring this retransmit (PRACK active)\n");
+ return;
+ } else if (res == -3) {
+ append_history(p, "PrIgnore", "Ignoring this response - out of order (PRACK active)\n");
+ return;
+ }
+ }
+ if (activeextensions & SIP_OPT_TIMER) {
+ ast_debug(3, "!=!=!=!=!=! The other side activated Session timers! \n");
+ /* Do nothing, funny enough */
+ }
+ }
+
+ 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) {
@@ -23612,6 +23855,9 @@
} else if (sipmethod == SIP_INFO) {
/* More good gravy! */
handle_response_info(p, resp, rest, req, seqno);
+ } else if (sipmethod == SIP_PRACK) {
+ /* More good candy! */
+ handle_response_prack(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_MESSAGE) {
/* More good gravy! */
handle_response_message(p, resp, rest, req, seqno);
@@ -24771,6 +25017,37 @@
return 0;
}
+/*! Support for the SIP Prack method
+ */
+static int handle_request_prack(struct sip_pvt *p, struct sip_request *req)
+{
+ const char *rack = sip_get_header(req, "RAck");
+ uint32_t rseq, cseq;
+
+ if(sscanf(rack, "%30u %30u", &rseq, &cseq) != 2) {
+ /* we did not get proper rseq/cseq */
+ transmit_response(p, "481 Could not get proper rseq/cseq in Rack", req);
+ }
+ ast_debug(3, "!=!=!=!=!=!= Got PRACK with rseq %u and cseq %u \n", rseq, cseq);
+ if (rseq <= p->rseq) {
+ /* Ack the retransmits */
+ int acked = __sip_ack(p, cseq, 1 /* response */, 0, rseq);
+ ast_debug(2, "!=!=!=!=!=! Tried acking the response - %s \n", acked ? "Sucess" : "Total utterly failure");
+ }
+ append_history(p, "PRACK", "PRACK received Rseq %u", rseq);
+ transmit_response(p, "200 OK", req);
+ if (ast_test_flag(&p->flags[2], SIP_PAGE3_ANSWER_WAIT_FOR_PRACK)) {
+ /* If the response sent reliably contained an SDP, we're not allowed to answer
+ until we have a PRACK response
+ */
+ ast_debug(2, "-<-<--<-<-<-<- Finally a good time to answer call (PRACK arrived) %s \n", ast_channel_name(p->owner));
+ ast_clear_flag(&p->flags[2], SIP_PAGE3_ANSWER_WAIT_FOR_PRACK);
+ sip_answer(p->owner);
+ }
+ ast_clear_flag(&p->flags[2], SIP_PAGE3_INVITE_WAIT_FOR_PRACK); /* Clear flag */
+ return 0;
+}
+
/*!
* \brief Handle incoming INVITE request
* \note If the INVITE has a Replaces header, it is part of an
@@ -24841,6 +25118,23 @@
Include the Require: option tags for further processing as well */
p->sipoptions |= required_profile;
p->reqsipoptions = required_profile;
+
+ /* Check if the request supports or require PRACK */
+ 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);
+ p->invitestate = INV_COMPLETED;
+ if (!p->lastinvite) {
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
+ }
+ res = 0;
+ goto request_invite_cleanup;
+ }
+ }
/* 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) && ast_channel_state(p->owner) != AST_STATE_UP) {
@@ -24904,7 +25198,7 @@
* 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.
*/
- __sip_ack(p, p->pendinginvite, 1, 0);
+ __sip_ack(p, p->pendinginvite, 1, 0, 0);
} else {
/* We already have a pending invite. Sorry. You are on hold. */
p->glareinvite = seqno;
@@ -27234,7 +27528,7 @@
return 0;
} else if (auth_result == AUTH_SUCCESSFUL && p->lastinvite) {
/* We need to stop retransmitting the 401 */
- __sip_ack(p, p->lastinvite, 1, 0);
+ __sip_ack(p, p->lastinvite, 1, 0, 0);
}
publish_type = determine_sip_publish_type(req, event, etag, expires_str, &expires_int);
@@ -28115,12 +28409,15 @@
case SIP_UPDATE:
res = handle_request_update(p, req);
break;
+ case SIP_PRACK:
+ res = handle_request_prack(p, req);
+ break;
case SIP_ACK:
/* Make sure we don't ignore this */
if (seqno == p->pendinginvite) {
p->invitestate = INV_TERMINATED;
p->pendinginvite = 0;
- acked = __sip_ack(p, seqno, 1 /* response */, 0);
+ acked = __sip_ack(p, seqno, 1 /* response */, 0, 0);
if (find_sdp(req)) {
if (process_sdp(p, req, SDP_T38_NONE)) {
return -1;
@@ -28133,7 +28430,7 @@
} else if (p->glareinvite == seqno) {
/* handle ack for the 491 pending sent for glareinvite */
p->glareinvite = 0;
- acked = __sip_ack(p, seqno, 1, 0);
+ acked = __sip_ack(p, seqno, 1, 0, 0);
}
if (!acked) {
/* Got an ACK that did not match anything. Ignore
@@ -29895,6 +30192,9 @@
} 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);
+ } else if (!strcasecmp(v->name, "prack")) {
+ ast_set_flag(&mask[2], SIP_PAGE3_PRACK);
+ ast_set2_flag(&flags[2], ast_true(v->value), SIP_PAGE3_PRACK);
} else
res = 0;
@@ -31165,6 +31465,7 @@
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;
@@ -31537,6 +31838,8 @@
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")) {
Modified: team/oej/darjeeling-prack-11/channels/sip/include/dialog.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-11/channels/sip/include/dialog.h?view=diff&rev=383212&r1=383211&r2=383212
==============================================================================
--- team/oej/darjeeling-prack-11/channels/sip/include/dialog.h (original)
+++ team/oej/darjeeling-prack-11/channels/sip/include/dialog.h Fri Mar 15 09:07:02 2013
@@ -67,7 +67,7 @@
/*! \brief Acknowledges receipt of a packet and stops retransmission
* called with p locked*/
-int __sip_ack(struct sip_pvt *p, uint32_t seqno, int resp, int sipmethod);
+int __sip_ack(struct sip_pvt *p, uint32_t seqno, int resp, int sipmethod, uint32_t rseqno);
/*! \brief Pretend to ack all packets
* called with p locked */
Modified: team/oej/darjeeling-prack-11/channels/sip/include/reqresp_parser.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-11/channels/sip/include/reqresp_parser.h?view=diff&rev=383212&r1=383211&r2=383212
==============================================================================
--- team/oej/darjeeling-prack-11/channels/sip/include/reqresp_parser.h (original)
+++ team/oej/darjeeling-prack-11/channels/sip/include/reqresp_parser.h Fri Mar 15 09:07:02 2013
@@ -154,6 +154,14 @@
* \param unsupported out buffer length (optional)
*/
unsigned int parse_sip_options(const char *options, char *unsupported, size_t unsupported_len);
+
+/*!
+ * \brief Parse required header in incoming packet or response
+ * returns bitmap
+ *
+ * \param option list
+ */
+unsigned int parse_required_sip_options(const char *options);
/*!
* \brief Compare two URIs as described in RFC 3261 Section 19.1.4
Modified: team/oej/darjeeling-prack-11/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/darjeeling-prack-11/channels/sip/include/sip.h?view=diff&rev=383212&r1=383211&r2=383212
==============================================================================
--- team/oej/darjeeling-prack-11/channels/sip/include/sip.h (original)
+++ team/oej/darjeeling-prack-11/channels/sip/include/sip.h Fri Mar 15 09:07:02 2013
@@ -161,7 +161,7 @@
* \todo This string should be set dynamically. We only support REFER and SUBSCRIBE if we have
* allowsubscribe and allowrefer on in sip.conf.
*/
-#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH"
+#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, PRACK"
/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_SIP_PORT 5060
@@ -194,6 +194,7 @@
#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 */
@@ -234,6 +235,7 @@
#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
#define DEFAULT_STORE_SIP_CAUSE FALSE /*!< Don't store HASH(SIP_CAUSE,<channel name>) for channels by default */
#endif
+#define DEFAULT_PRACK FALSE /*!< Default: Prack is turned off */
/*@}*/
/*! \name SIPflags
@@ -374,10 +376,14 @@
#define SIP_PAGE3_DIRECT_MEDIA_OUTGOING (1 << 4) /*!< DP: Only send direct media reinvites on outgoing calls */
#define SIP_PAGE3_USE_AVPF (1 << 5) /*!< DGP: Support a minimal AVPF-compatible profile */
#define SIP_PAGE3_ICE_SUPPORT (1 << 6) /*!< DGP: Enable ICE support */
+#define SIP_PAGE3_PRACK (1 << 7) /*!< DPG: Allow snom aoc messages */
+#define SIP_PAGE3_100REL (1 << 8) /*!< D: If PRACK is active for a specific dialog */
+#define SIP_PAGE3_INVITE_WAIT_FOR_PRACK (1 << 9) /*!< D: Wait for PRACK response before sending 200 OK */
[... 201 lines stripped ...]
More information about the svn-commits
mailing list