Major RTP fixes for using inbound SDP on outbound connection, get rid of
old local rtp stuff... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
parent
93d1d0dcfb
commit
0d32a85be1
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@ -53,6 +53,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/utils.h"
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#include "asterisk/app.h"
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#include "asterisk/causes.h"
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#include "asterisk/rtp.h"
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#include "asterisk/manager.h"
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#include "asterisk/privacy.h"
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@ -310,7 +311,7 @@ static void hanguptree(struct localuser *outgoing, struct ast_channel *exception
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} while (0)
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static int onedigit_goto(struct ast_channel *chan, char *context, char exten, int pri)
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static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
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{
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char rexten[2] = { exten, '\0' };
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@ -380,7 +381,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
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int pos;
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int single;
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struct ast_channel *winner;
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char *context = NULL;
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const char *context = NULL;
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char cidname[AST_MAX_EXTENSION];
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single = (outgoing && !outgoing->next && !ast_test_flag(outgoing, OPT_MUSICBACK | OPT_RINGBACK));
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@ -475,6 +476,7 @@ static struct ast_channel *wait_for_answer(struct ast_channel *in, struct localu
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ast_clear_flag(o, DIAL_STILLGOING);
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HANDLE_CAUSE(cause, in);
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} else {
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ast_rtp_make_compatible(o->chan, in);
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if (o->chan->cid.cid_num)
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free(o->chan->cid.cid_num);
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o->chan->cid.cid_num = NULL;
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@ -744,16 +746,17 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
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long timelimit = 0;
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long play_warning = 0;
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long warning_freq=0;
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char *warning_sound=NULL;
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char *end_sound=NULL;
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char *start_sound=NULL;
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const char *warning_sound=NULL;
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const char *end_sound=NULL;
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const char *start_sound=NULL;
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char *dtmfcalled=NULL, *dtmfcalling=NULL;
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char *var;
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const char *var;
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char status[256];
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int play_to_caller=0,play_to_callee=0;
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int sentringing=0, moh=0;
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char *outbound_group = NULL;
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char *macro_result = NULL, *macro_transfer_dest = NULL;
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const char *outbound_group = NULL;
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const char *macro_result = NULL;
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char *macro_transfer_dest = NULL;
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int digit = 0, result = 0;
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time_t start_time, answer_time, end_time;
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struct ast_app *app = NULL;
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@ -1052,6 +1055,9 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
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}
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}
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/* Setup outgoing SDP to match incoming one */
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ast_rtp_make_compatible(tmp->chan, chan);
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/* Inherit specially named variables from parent channel */
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ast_channel_inherit_variables(chan, tmp->chan);
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@ -1190,7 +1196,7 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
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if (peer->name)
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pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
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number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
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number = (char *)pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
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if (!number)
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number = numsubst;
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pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
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@ -1602,7 +1608,8 @@ static int dial_exec(struct ast_channel *chan, void *data)
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static int retrydial_exec(struct ast_channel *chan, void *data)
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{
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char *announce = NULL, *context = NULL, *dialdata = NULL;
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char *announce = NULL, *dialdata = NULL;
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const char *context = NULL;
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int sleep = 0, loops = 0, res = 0;
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struct localuser *u;
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struct ast_flags peerflags;
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@ -2095,7 +2095,6 @@ static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
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add_header(&resp, "X", sub->txident);
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add_header(&resp, "I", sub->cxident);
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/*add_header(&resp, "S", "");*/
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ast_rtp_offered_from_local(sub->rtp, 0);
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add_sdp(&resp, sub, rtp);
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/* SC: fill in new fields */
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resp.cmd = MGCP_CMD_MDCX;
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@ -2129,7 +2128,6 @@ static int transmit_connect_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
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/* SC: X header should not be sent. kept for compatibility */
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add_header(&resp, "X", sub->txident);
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/*add_header(&resp, "S", "");*/
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ast_rtp_offered_from_local(sub->rtp, 1);
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add_sdp(&resp, sub, rtp);
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/* SC: fill in new fields */
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resp.cmd = MGCP_CMD_CRCX;
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@ -3948,7 +3946,7 @@ static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, stru
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/* XXX Is there such thing as video support with MGCP? XXX */
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struct mgcp_subchannel *sub;
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sub = chan->tech_pvt;
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if (sub) {
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if (sub && !sub->alreadygone) {
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transmit_modify_with_sdp(sub, rtp, codecs);
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return 0;
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}
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@ -4509,7 +4509,6 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
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}
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respprep(&resp, p, msg, req);
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if (p->rtp) {
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ast_rtp_offered_from_local(p->rtp, 0);
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add_sdp(&resp, p);
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} else {
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ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
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@ -4583,7 +4582,6 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
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add_header(&req, "Allow", ALLOWED_METHODS);
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if (sipdebug)
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add_header(&req, "X-asterisk-info", "SIP re-invite (RTP bridge)");
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ast_rtp_offered_from_local(p->rtp, 1);
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add_sdp(&req, p);
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/* Use this as the basis */
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copy_request(&p->initreq, &req);
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@ -4924,7 +4922,6 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
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}
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}
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if (sdp && p->rtp) {
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ast_rtp_offered_from_local(p->rtp, 1);
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add_sdp(&req, p);
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} else {
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add_header_contentLength(&req, 0);
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@ -12691,6 +12688,11 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
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if (!p)
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return -1;
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ast_mutex_lock(&p->lock);
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if (ast_test_flag(p, SIP_ALREADYGONE)) {
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/* If we're destroyed, don't bother */
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ast_mutex_unlock(&p->lock);
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return 0;
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}
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if (rtp)
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ast_rtp_get_peer(rtp, &p->redirip);
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else
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25
frame.c
25
frame.c
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@ -1299,3 +1299,28 @@ int ast_frame_slinear_sum(struct ast_frame *f1, struct ast_frame *f2)
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return 0;
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}
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struct ast_frame *ast_frame_enqueue(struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe)
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{
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struct ast_frame *cur, *oldhead;
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int len=0;
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if (f && dupe)
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f = ast_frdup(f);
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if (!f)
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return head;
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f->next = NULL;
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if (!head)
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return f;
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cur = head;
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while(cur->next) {
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cur = cur->next;
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len++;
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if (len >= maxlen) {
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oldhead = head;
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head = head->next;
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ast_frfree(oldhead);
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}
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}
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return head;
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}
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@ -452,6 +452,10 @@ extern int ast_codec_get_samples(struct ast_frame *f);
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/*! \brief Returns the number of bytes for the number of samples of the given format */
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extern int ast_codec_get_len(int format, int samples);
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/*! \brief Appends a frame to the end of a list of frames, truncating the maximum length of the list */
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extern struct ast_frame *ast_frame_enqueue(struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe);
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/*! \brief Gets duration in ms of interpolation frame for a format */
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static inline int ast_codec_interp_len(int format)
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{
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@ -135,7 +135,6 @@ void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
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/* Mapping between RTP payload format codes and Asterisk codes: */
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struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt);
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int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code);
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void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local);
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void ast_rtp_get_current_formats(struct ast_rtp* rtp,
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int* astFormats, int* nonAstFormats);
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@ -154,6 +153,8 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
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void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
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int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src);
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void ast_rtp_stop(struct ast_rtp *rtp);
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void ast_rtp_init(void);
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127
rtp.c
127
rtp.c
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@ -124,7 +124,6 @@ struct ast_rtp {
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int rtp_lookup_code_cache_isAstFormat;
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int rtp_lookup_code_cache_code;
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int rtp_lookup_code_cache_result;
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int rtp_offered_from_local;
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struct ast_rtcp *rtcp;
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};
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@ -724,10 +723,98 @@ void ast_rtp_pt_default(struct ast_rtp* rtp)
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rtp->rtp_lookup_code_cache_result = 0;
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}
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static void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src)
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{
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int i;
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/* Copy payload types from source to destination */
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for (i=0; i < MAX_RTP_PT; ++i) {
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dest->current_RTP_PT[i].isAstFormat =
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src->current_RTP_PT[i].isAstFormat;
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dest->current_RTP_PT[i].code =
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src->current_RTP_PT[i].code;
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}
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dest->rtp_lookup_code_cache_isAstFormat = 0;
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dest->rtp_lookup_code_cache_code = 0;
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dest->rtp_lookup_code_cache_result = 0;
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}
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/*--- get_proto: Get channel driver interface structure */
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static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
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{
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struct ast_rtp_protocol *cur;
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cur = protos;
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while(cur) {
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if (cur->type == chan->type) {
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return cur;
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}
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cur = cur->next;
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}
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return NULL;
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}
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int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
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{
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struct ast_rtp *destp, *srcp; /* Audio RTP Channels */
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struct ast_rtp *vdestp, *vsrcp; /* Video RTP channels */
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struct ast_rtp_protocol *destpr, *srcpr;
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/* Lock channels */
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ast_mutex_lock(&dest->lock);
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while(ast_mutex_trylock(&src->lock)) {
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ast_mutex_unlock(&dest->lock);
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usleep(1);
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ast_mutex_lock(&dest->lock);
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}
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/* Find channel driver interfaces */
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destpr = get_proto(dest);
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srcpr = get_proto(src);
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if (!destpr) {
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ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
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ast_mutex_unlock(&dest->lock);
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ast_mutex_unlock(&src->lock);
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return 0;
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}
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if (!srcpr) {
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ast_log(LOG_WARNING, "Channel '%s' has no RTP, not doing anything\n", src->name);
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ast_mutex_unlock(&dest->lock);
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ast_mutex_unlock(&src->lock);
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return 0;
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}
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/* Get audio and video interface (if native bridge is possible) */
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destp = destpr->get_rtp_info(dest);
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if (destpr->get_vrtp_info)
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vdestp = destpr->get_vrtp_info(dest);
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else
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vdestp = NULL;
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srcp = srcpr->get_rtp_info(src);
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if (srcpr->get_vrtp_info)
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vsrcp = srcpr->get_vrtp_info(src);
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else
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vsrcp = NULL;
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/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
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if (!destp || !srcp) {
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/* Somebody doesn't want to play... */
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ast_mutex_unlock(&dest->lock);
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ast_mutex_unlock(&src->lock);
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return 0;
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}
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ast_rtp_pt_copy(destp, srcp);
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if (vdestp && vsrcp)
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ast_rtp_pt_copy(vdestp, vsrcp);
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ast_mutex_unlock(&dest->lock);
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ast_mutex_unlock(&src->lock);
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ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
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return 1;
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}
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/* Make a note of a RTP paymoad type that was seen in a SDP "m=" line. */
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/* By default, use the well-known value for this type (although it may */
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/* still be set to a different value by a subsequent "a=rtpmap:" line): */
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void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) {
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void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
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{
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if (pt < 0 || pt > MAX_RTP_PT)
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return; /* bogus payload type */
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@ -739,7 +826,9 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) {
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/* Make a note of a RTP payload type (with MIME type) that was seen in */
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/* a SDP "a=rtpmap:" line. */
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void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
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char* mimeType, char* mimeSubtype) {
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char* mimeType, char* mimeSubtype)
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{
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int i;
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if (pt < 0 || pt > MAX_RTP_PT)
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@ -770,13 +859,6 @@ void ast_rtp_get_current_formats(struct ast_rtp* rtp,
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}
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}
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void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local) {
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if (rtp)
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rtp->rtp_offered_from_local = local;
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else
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ast_log(LOG_WARNING, "rtp structure is null\n");
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}
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struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt)
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{
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struct rtpPayloadType result;
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@ -786,8 +868,7 @@ struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt)
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return result; /* bogus payload type */
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/* Start with the negotiated codecs */
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if (!rtp->rtp_offered_from_local)
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result = rtp->current_RTP_PT[pt];
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result = rtp->current_RTP_PT[pt];
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/* If it doesn't exist, check our static RTP type list, just in case */
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if (!result.code)
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@ -829,7 +910,8 @@ int ast_rtp_lookup_code(struct ast_rtp* rtp, const int isAstFormat, const int co
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return -1;
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}
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char* ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code) {
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char* ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code)
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{
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int i;
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@ -1485,21 +1567,6 @@ int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
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return 0;
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}
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/*--- get_proto: Get channel driver interface structure */
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static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
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{
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struct ast_rtp_protocol *cur;
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cur = protos;
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while(cur) {
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if (cur->type == chan->type) {
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return cur;
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}
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cur = cur->next;
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}
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return NULL;
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}
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/* ast_rtp_bridge: Bridge calls. If possible and allowed, initiate
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re-invite so the peers exchange media directly outside
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of Asterisk. */
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@ -1698,11 +1765,11 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
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*rc = who;
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if (option_debug)
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ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
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if ((c0->tech_pvt == pvt0) && (!c0->_softhangup)) {
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if ((c0->tech_pvt == pvt0)) {
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if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
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ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
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}
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if ((c1->tech_pvt == pvt1) && (!c1->_softhangup)) {
|
||||
if ((c1->tech_pvt == pvt1)) {
|
||||
if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
|
||||
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in New Issue