Don't duplicate function if not needed...
- 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
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1 changed files with 14 additions and 31 deletions
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@ -1221,7 +1221,7 @@ static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *r
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static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
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static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch);
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static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init);
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static int transmit_reinvite_with_sdp(struct sip_pvt *p);
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static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version);
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static int transmit_info_with_digit(struct sip_pvt *p, const char digit);
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static int transmit_info_with_vidupdate(struct sip_pvt *p);
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static int transmit_message_with_text(struct sip_pvt *p, const char *text);
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@ -1520,7 +1520,6 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
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/*------ T38 Support --------- */
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static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt, int reinvite); /*!< T38 negotiation helper function */
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static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
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static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p);
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static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
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static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
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@ -4120,7 +4119,7 @@ static struct ast_frame *sip_read(struct ast_channel *ast)
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if (option_debug > 2)
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ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name);
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p->t38.state = T38_LOCAL_REINVITE;
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transmit_reinvite_with_t38_sdp(p);
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transmit_reinvite_with_sdp(p, TRUE);
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if (option_debug > 1)
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ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, ast->name);
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}
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@ -6402,8 +6401,11 @@ static int determine_firstline_parts(struct sip_request *req)
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INVITE that opened the SIP dialogue
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We reinvite so that the audio stream (RTP) go directly between
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the SIP UAs. SIP Signalling stays with * in the path.
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If t38version is TRUE, we send T38 SDP for re-invite from audio/video to
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T38 UDPTL transmission on the channel
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*/
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static int transmit_reinvite_with_sdp(struct sip_pvt *p)
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static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version)
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{
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struct sip_request req;
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@ -6415,29 +6417,10 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
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add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
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if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
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append_history(p, "ReInv", "Re-invite sent");
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add_sdp(&req, p);
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/* Use this as the basis */
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initialize_initreq(p, &req);
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p->lastinvite = p->ocseq;
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return send_request(p, &req, XMIT_CRITICAL, p->ocseq);
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}
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/*! \brief Transmit reinvite with T38 SDP
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We reinvite so that the T38 processing can take place.
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SIP Signalling stays with * in the path.
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*/
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static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p)
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{
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struct sip_request req;
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reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);
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add_header(&req, "Allow", ALLOWED_METHODS);
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add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
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if (sipdebug)
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add_header(&req, "X-asterisk-info", "SIP re-invite (T38 switchover)");
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ast_udptl_offered_from_local(p->udptl, 1);
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if (t38version)
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add_t38_sdp(&req, p);
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else
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add_sdp(&req, p);
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/* Use this as the basis */
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initialize_initreq(p, &req);
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p->lastinvite = p->ocseq;
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@ -11564,7 +11547,7 @@ static void check_pendings(struct sip_pvt *p)
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if (option_debug)
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ast_log(LOG_DEBUG, "Sending pending reinvite on '%s'\n", p->callid);
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/* Didn't get to reinvite yet, so do it now */
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transmit_reinvite_with_sdp(p);
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transmit_reinvite_with_sdp(p, FALSE);
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ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
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}
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}
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@ -16479,7 +16462,7 @@ static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
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if (option_debug > 2) {
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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);
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}
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transmit_reinvite_with_t38_sdp(p);
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transmit_reinvite_with_sdp(p, TRUE);
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} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
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if (option_debug > 2) {
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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);
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@ -16524,7 +16507,7 @@ static int sip_handle_t38_reinvite(struct ast_channel *chan, struct sip_pvt *pvt
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else
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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));
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}
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transmit_reinvite_with_t38_sdp(p);
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transmit_reinvite_with_sdp(p, TRUE);
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} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
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if (option_debug > 2) {
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if (flag)
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@ -16668,7 +16651,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
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if (option_debug > 2) {
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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));
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}
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transmit_reinvite_with_sdp(p);
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transmit_reinvite_with_sdp(p, FALSE);
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} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
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if (option_debug > 2) {
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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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