[svn-commits] oej: trunk r46385 - /trunk/channels/chan_sip.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Sat Oct 28 12:30:31 MST 2006
Author: oej
Date: Sat Oct 28 14:30:31 2006
New Revision: 46385
URL: http://svn.digium.com/view/asterisk?rev=46385&view=rev
Log:
Don't duplicate function if not needed...
- removing transmit_reinvite_with_t38_sdp in favour of adding an argument to
transmit_reinvite_with_sdp
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=46385&r1=46384&r2=46385&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sat Oct 28 14:30:31 2006
@@ -1221,7 +1221,7 @@
static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch);
static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init);
-static int transmit_reinvite_with_sdp(struct sip_pvt *p);
+static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version);
static int transmit_info_with_digit(struct sip_pvt *p, const char digit);
static int transmit_info_with_vidupdate(struct sip_pvt *p);
static int transmit_message_with_text(struct sip_pvt *p, const char *text);
@@ -1520,7 +1520,6 @@
/*------ T38 Support --------- */
static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */
static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
-static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p);
static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
@@ -4120,7 +4119,7 @@
if (option_debug > 2)
ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name);
p->t38.state = T38_LOCAL_REINVITE;
- transmit_reinvite_with_t38_sdp(p);
+ transmit_reinvite_with_sdp(p, TRUE);
if (option_debug > 1)
ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, ast->name);
}
@@ -6402,8 +6401,11 @@
INVITE that opened the SIP dialogue
We reinvite so that the audio stream (RTP) go directly between
the SIP UAs. SIP Signalling stays with * in the path.
+
+ If t38version is TRUE, we send T38 SDP for re-invite from audio/video to
+ T38 UDPTL transmission on the channel
*/
-static int transmit_reinvite_with_sdp(struct sip_pvt *p)
+static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version)
{
struct sip_request req;
@@ -6415,29 +6417,10 @@
add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "ReInv", "Re-invite sent");
- add_sdp(&req, p);
- /* Use this as the basis */
- initialize_initreq(p, &req);
- p->lastinvite = p->ocseq;
- return send_request(p, &req, XMIT_CRITICAL, p->ocseq);
-}
-
-/*! \brief Transmit reinvite with T38 SDP
- We reinvite so that the T38 processing can take place.
- SIP Signalling stays with * in the path.
-*/
-static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p)
-{
- struct sip_request req;
-
- reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);
-
- add_header(&req, "Allow", ALLOWED_METHODS);
- add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
- if (sipdebug)
- add_header(&req, "X-asterisk-info", "SIP re-invite (T38 switchover)");
- ast_udptl_offered_from_local(p->udptl, 1);
- add_t38_sdp(&req, p);
+ if (t38version)
+ add_t38_sdp(&req, p);
+ else
+ add_sdp(&req, p);
/* Use this as the basis */
initialize_initreq(p, &req);
p->lastinvite = p->ocseq;
@@ -11564,7 +11547,7 @@
if (option_debug)
ast_log(LOG_DEBUG, "Sending pending reinvite on '%s'\n", p->callid);
/* Didn't get to reinvite yet, so do it now */
- transmit_reinvite_with_sdp(p);
+ transmit_reinvite_with_sdp(p, FALSE);
ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
}
}
@@ -16479,7 +16462,7 @@
if (option_debug > 2) {
ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
}
- transmit_reinvite_with_t38_sdp(p);
+ transmit_reinvite_with_sdp(p, TRUE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
if (option_debug > 2) {
ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
@@ -16524,7 +16507,7 @@
else
ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(p->ourip));
}
- transmit_reinvite_with_t38_sdp(p);
+ transmit_reinvite_with_sdp(p, TRUE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
if (option_debug > 2) {
if (flag)
@@ -16668,7 +16651,7 @@
if (option_debug > 2) {
ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip));
}
- transmit_reinvite_with_sdp(p);
+ transmit_reinvite_with_sdp(p, FALSE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
if (option_debug > 2) {
ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip));
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