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Don't duplicate function if not needed...

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- removing transmit_reinvite_with_t38_sdp in favour of adding an argument to 
  transmit_reinvite_with_sdp


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2006-10-28 19:30:31 +00:00
parent 29e489e000
commit 336a4602ba
1 changed files with 14 additions and 31 deletions
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45
channels/chan_sip.c
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@ -1221,7 +1221,7 @@ static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *r
static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch); 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_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_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_digit(struct sip_pvt *p, const char digit);
static int transmit_info_with_vidupdate(struct sip_pvt *p); static int transmit_info_with_vidupdate(struct sip_pvt *p);
static int transmit_message_with_text(struct sip_pvt *p, const char *text); static int transmit_message_with_text(struct sip_pvt *p, const char *text);
@ -1520,7 +1520,6 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
/*------ T38 Support --------- */ /*------ T38 Support --------- */
static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */ 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_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 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); static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
@ -4120,7 +4119,7 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
if (option_debug > 2) if (option_debug > 2)
ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name); ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name);
p->t38.state = T38_LOCAL_REINVITE; p->t38.state = T38_LOCAL_REINVITE;
transmit_reinvite_with_t38_sdp(p); transmit_reinvite_with_sdp(p, TRUE);
if (option_debug > 1) if (option_debug > 1)
ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, ast->name); ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, ast->name);
} }
@ -6402,8 +6401,11 @@ static int determine_firstline_parts(struct sip_request *req)
INVITE that opened the SIP dialogue INVITE that opened the SIP dialogue
We reinvite so that the audio stream (RTP) go directly between We reinvite so that the audio stream (RTP) go directly between
the SIP UAs. SIP Signalling stays with * in the path. 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; struct sip_request req;
@ -6415,29 +6417,10 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)"); add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "ReInv", "Re-invite sent"); append_history(p, "ReInv", "Re-invite sent");
add_sdp(&req, p); if (t38version)
/* Use this as the basis */ add_t38_sdp(&req, p);
initialize_initreq(p, &req); else
p->lastinvite = p->ocseq; add_sdp(&req, p);
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);
/* Use this as the basis */ /* Use this as the basis */
initialize_initreq(p, &req); initialize_initreq(p, &req);
p->lastinvite = p->ocseq; p->lastinvite = p->ocseq;
@ -11564,7 +11547,7 @@ static void check_pendings(struct sip_pvt *p)
if (option_debug) if (option_debug)
ast_log(LOG_DEBUG, "Sending pending reinvite on '%s'\n", p->callid); ast_log(LOG_DEBUG, "Sending pending reinvite on '%s'\n", p->callid);
/* Didn't get to reinvite yet, so do it now */ /* 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); ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
} }
} }
@ -16479,7 +16462,7 @@ static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
if (option_debug > 2) { 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); 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)) { } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
if (option_debug > 2) { 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); 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 @@ static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt
else 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)); 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)) { } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
if (option_debug > 2) { if (option_debug > 2) {
if (flag) if (flag)
@ -16668,7 +16651,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
if (option_debug > 2) { 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)); 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)) { } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
if (option_debug > 2) { 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)); 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));