[asterisk-commits] oej: branch oej/pinetestedition-1.8 r383034 - in /team/oej/pinetestedition-1....
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Mar 13 11:29:48 CDT 2013
Author: oej
Date: Wed Mar 13 11:29:44 2013
New Revision: 383034
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=383034
Log:
Adding darjeeling and half of the rtcp support. Still working on merging that.
Added:
team/oej/pinetestedition-1.8/patches/darjeeling-prack-1.8.diff (with props)
Modified:
team/oej/pinetestedition-1.8/CREDITS
team/oej/pinetestedition-1.8/channels/chan_sip.c
team/oej/pinetestedition-1.8/channels/sip/include/dialog.h
team/oej/pinetestedition-1.8/channels/sip/include/reqresp_parser.h
team/oej/pinetestedition-1.8/channels/sip/include/sip.h
team/oej/pinetestedition-1.8/channels/sip/reqresp_parser.c
team/oej/pinetestedition-1.8/configs/sip.conf.sample
Modified: team/oej/pinetestedition-1.8/CREDITS
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinetestedition-1.8/CREDITS?view=diff&rev=383034&r1=383033&r2=383034
==============================================================================
--- team/oej/pinetestedition-1.8/CREDITS (original)
+++ team/oej/pinetestedition-1.8/CREDITS Wed Mar 13 11:29:44 2013
@@ -25,6 +25,8 @@
Nordicom Norge AS, Kristiansand, Norway, for funding work with RTCP support
and Call Quality Records.
+
+Nordicom Norge AS for funding of the SIP PRACK support in chan_sip.
=== WISHLIST CONTRIBUTERS ===
Jeremy McNamara - SpeeX support
Modified: team/oej/pinetestedition-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinetestedition-1.8/channels/chan_sip.c?view=diff&rev=383034&r1=383033&r2=383034
==============================================================================
--- team/oej/pinetestedition-1.8/channels/chan_sip.c (original)
+++ team/oej/pinetestedition-1.8/channels/chan_sip.c Wed Mar 13 11:29:44 2013
@@ -1245,10 +1245,10 @@
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);
@@ -1269,6 +1269,7 @@
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);
@@ -4125,6 +4126,9 @@
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;
@@ -4306,7 +4310,7 @@
/*! \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 */
@@ -4325,6 +4329,10 @@
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";
@@ -4336,6 +4344,7 @@
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
@@ -4366,8 +4375,8 @@
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;
}
@@ -4385,7 +4394,7 @@
}
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);
}
}
@@ -4498,6 +4507,10 @@
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;
@@ -4526,10 +4539,35 @@
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);
@@ -4555,7 +4593,7 @@
}
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) {
@@ -6939,7 +6977,13 @@
struct sip_pvt *p = ast->tech_pvt;
sip_pvt_lock(p);
- if (ast->_state != AST_STATE_UP) {
+ if (ast->_state != 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->name);
+ sip_pvt_unlock(p);
+ return 0;
+ }
+ if (ast->_state != 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);
@@ -10401,13 +10445,15 @@
* is supported for this dialog. */
static int add_supported_header(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 */
@@ -10925,7 +10971,6 @@
seqno = p->ocseq;
}
- /* A CANCEL must have the same branch as the INVITE that it is canceling. */
if (sipmethod == SIP_CANCEL) {
p->branch = p->invite_branch;
build_via(p);
@@ -11048,8 +11093,10 @@
{
struct sip_request resp;
uint32_t seqno = 0;
-
- if (reliable && (sscanf(get_header(req, "CSeq"), "%30u ", &seqno) != 1)) {
+ int res;
+
+ res = sscanf(get_header(req, "CSeq"), "%30u ", &seqno);
+ if (reliable && res != 1) {
ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq"));
return -1;
}
@@ -11060,6 +11107,10 @@
&& (!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);
@@ -11100,6 +11151,10 @@
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);
}
@@ -11112,6 +11167,7 @@
}
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);
}
@@ -11199,6 +11255,7 @@
respprep(&resp, p, msg, req);
append_date(&resp);
add_header(&resp, "Unsupported", unsupported);
+ add_required_respheader(&resp);
return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
@@ -11252,6 +11309,7 @@
struct sip_request resp;
respprep(&resp, p, msg, req);
append_date(&resp);
+ add_required_respheader(&resp);
return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
@@ -11261,6 +11319,7 @@
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);
}
@@ -11273,6 +11332,7 @@
snprintf(tmp, sizeof(tmp), "%d", min_expiry);
respprep(&resp, p, msg, req);
add_header(&resp, "Min-Expires", tmp);
+ add_required_respheader(&resp);
return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
@@ -12359,6 +12419,18 @@
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);
}
@@ -12769,7 +12841,69 @@
}
/*!
- * \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
@@ -12784,7 +12918,9 @@
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) {
@@ -12820,12 +12956,19 @@
}
snprintf(buf, sizeof(buf), "%d", p->expiry);
add_header(&req, "Expires", buf);
+ } else if (sipmethod == SIP_PRACK) {
+ /* Place holder */
+ /* 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");
}
@@ -18341,6 +18484,7 @@
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, " Comfort Noise: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOW_CN)));
@@ -18481,6 +18625,7 @@
/* - 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));
@@ -19080,7 +19225,9 @@
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");
@@ -19463,6 +19610,8 @@
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 Transfer mode: %s\n", transfermode2str(cur->allowtransfer));
+ 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, " 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,
@@ -20716,6 +20865,48 @@
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", 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
*
@@ -20972,7 +21163,7 @@
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);
@@ -20981,6 +21172,9 @@
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;
@@ -21044,13 +21238,16 @@
}
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);
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 {
@@ -21155,6 +21352,10 @@
/* Check for Session-Timers related headers */
if (st_get_mode(p, 0) != SESSION_TIMER_MODE_REFUSE && p->outgoing_call == TRUE && !reinvite) {
+ /* XXX Code should check in response if there's a "Require: timer"
+ header. If there is, sessions timer is enabled for this dialog
+ If not, only this side (UAC) do session timers.
+ */
p_hdrval = (char*)get_header(req, "Session-Expires");
if (!ast_strlen_zero(p_hdrval)) {
/* UAS supports Session-Timers */
@@ -21872,6 +22073,9 @@
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 */
@@ -21913,7 +22117,7 @@
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) {
@@ -21932,13 +22136,14 @@
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.
@@ -21957,6 +22162,49 @@
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 = 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(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");
+ }
+ }
+
+ 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) {
@@ -21975,6 +22223,9 @@
} 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);
@@ -23122,6 +23373,37 @@
return 0;
}
+/*! Support for the SIP Prack method
+ */
+static int handle_request_prack(struct sip_pvt *p, struct sip_request *req)
+{
+ const char *rack = 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", p->owner->name);
+ 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
@@ -23183,7 +23465,7 @@
p->invitestate = INV_COMPLETED;
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
- res = -1;
+ res = 0;
goto request_invite_cleanup;
}
}
@@ -23192,6 +23474,23 @@
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) && p->owner->_state != AST_STATE_UP) {
@@ -23255,7 +23554,7 @@
* 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;
@@ -25369,7 +25668,7 @@
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);
@@ -26199,12 +26498,15 @@
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;
@@ -26217,7 +26519,7 @@
} 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
@@ -27863,6 +28165,9 @@
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
} else if (!strcasecmp(v->name, "comfort-noise")) {
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOW_CN);
+ } 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;
@@ -29044,6 +29349,7 @@
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;
@@ -29384,6 +29690,8 @@
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/pinetestedition-1.8/channels/sip/include/dialog.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinetestedition-1.8/channels/sip/include/dialog.h?view=diff&rev=383034&r1=383033&r2=383034
==============================================================================
--- team/oej/pinetestedition-1.8/channels/sip/include/dialog.h (original)
+++ team/oej/pinetestedition-1.8/channels/sip/include/dialog.h Wed Mar 13 11:29:44 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/pinetestedition-1.8/channels/sip/include/reqresp_parser.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinetestedition-1.8/channels/sip/include/reqresp_parser.h?view=diff&rev=383034&r1=383033&r2=383034
==============================================================================
--- team/oej/pinetestedition-1.8/channels/sip/include/reqresp_parser.h (original)
+++ team/oej/pinetestedition-1.8/channels/sip/include/reqresp_parser.h Wed Mar 13 11:29:44 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/pinetestedition-1.8/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinetestedition-1.8/channels/sip/include/sip.h?view=diff&rev=383034&r1=383033&r2=383034
==============================================================================
--- team/oej/pinetestedition-1.8/channels/sip/include/sip.h (original)
+++ team/oej/pinetestedition-1.8/channels/sip/include/sip.h Wed Mar 13 11:29:44 2013
@@ -157,7 +157,7 @@
* \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
@@ -189,6 +189,7 @@
#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 */
@@ -225,6 +226,7 @@
#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
+#define DEFAULT_PRACK FALSE /*!< Default: Prack is turned off */
/*@}*/
/*! \name SIPflags
@@ -361,11 +363,15 @@
SIP_PAGE2_ALLOW_CN )
-#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
+#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
#define SIP_PAGE3_DIRECT_MEDIA_OUTGOING (1 << 1) /*!< DP: Only send direct media reinvites on outgoing calls */
+#define SIP_PAGE3_PRACK (1 << 2) /*!< DPG: Allow snom aoc messages */
+#define SIP_PAGE3_100REL (1 << 3) /*!< D: If PRACK is active for a specific dialog */
+#define SIP_PAGE3_INVITE_WAIT_FOR_PRACK (1 << 4) /*!< D: Wait for PRACK response before sending 200 OK */
+#define SIP_PAGE3_ANSWER_WAIT_FOR_PRACK (1 << 5) /*!< D: Send ANSWER when PRACK is received */
#define SIP_PAGE3_FLAGS_TO_COPY \
- (SIP_PAGE3_SNOM_AOC | SIP_PAGE3_DIRECT_MEDIA_OUTGOING)
+ (SIP_PAGE3_SNOM_AOC | SIP_PAGE3_PRACK | SIP_PAGE3_DIRECT_MEDIA_OUTGOING)
/*@}*/
@@ -409,6 +415,7 @@
* where the original response would be sent RELIABLE in an INVITE transaction
*/
enum xmittype {
+ XMIT_PRACK = 3, /*!< Transmit response the PRACK way: reliably, with re-transmits. */
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 */
@@ -572,7 +579,7 @@
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. */
- SIP_PRACK, /*!< Reliable pre-call signalling. Not supported in Asterisk. */
+ SIP_PRACK, /*!< Reliable pre-call signalling. */
SIP_BYE, /*!< End of a session */
SIP_REFER, /*!< Refer to another URI (transfer) */
SIP_SUBSCRIBE, /*!< Subscribe for updates (voicemail, session status, device status, presence) */
@@ -687,6 +694,7 @@
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 rtsave_path; /*!< G: Save path header on registration */
@@ -764,12 +772,14 @@
int headers; /*!< # of SIP Headers */
int method; /*!< Method of this request */
int lines; /*!< Body Content */
+ uint32_t rseqno; /*!< PRACK Rseq */
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 */
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;
@@ -976,6 +986,8 @@
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] */
@@ -1002,6 +1014,8 @@
uint32_t ocseq; /*!< Current outgoing seqno */
uint32_t icseq; /*!< Current incoming seqno */
uint32_t init_icseq; /*!< Initial incoming seqno from first request */
+ uint32_t rseq; /*!< Current PRACK rseq on our side*/
+ uint32_t irseq; /*!< Current PRACK rseq on their side*/
ast_group_t callgroup; /*!< Call group */
ast_group_t pickupgroup; /*!< Pickup group */
uint32_t lastinvite; /*!< Last seqno of invite */
@@ -1167,6 +1181,7 @@
int retrans; /*!< Retransmission number */
int method; /*!< SIP method for this packet */
uint32_t seqno; /*!< Sequence number */
+ uint32_t rseqno; /*!< PRACK Sequence number */
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 */
@@ -1778,7 +1793,7 @@
char * const text; /*!< Text id, as in standard */
} sip_options[] = { /* XXX used in 3 places */
/* RFC3262: PRACK 100% reliability */
- { SIP_OPT_100REL, NOT_SUPPORTED, "100rel" },
+ { SIP_OPT_100REL, SUPPORTED, "100rel" },
[... 3720 lines stripped ...]
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