Make RTP handle codecs (first pass)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
parent
fc7593e594
commit
cf57ba2310
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@ -1776,7 +1776,7 @@ static char *convertcap(int cap)
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}
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static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp)
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static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
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{
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/* XXX Deal with Video */
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struct oh323_pvt *p;
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@ -433,7 +433,7 @@ static int transmit_response(struct mgcp_subchannel *sub, char *msg, struct mgcp
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static int transmit_notify_request(struct mgcp_subchannel *sub, char *tone);
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static int transmit_modify_request(struct mgcp_subchannel *sub);
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static int transmit_notify_request_with_callerid(struct mgcp_subchannel *sub, char *tone, char *callerid);
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static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp);
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static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp, int codecs);
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static int transmit_connection_del(struct mgcp_subchannel *sub);
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static int transmit_audit_endpoint(struct mgcp_endpoint *p);
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static void start_rtp(struct mgcp_subchannel *sub);
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@ -1908,13 +1908,17 @@ static int add_sdp(struct mgcp_request *resp, struct mgcp_subchannel *sub, struc
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return 0;
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}
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static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp)
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static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp *rtp, int codecs)
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{
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struct mgcp_request resp;
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char local[256];
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char tmp[80];
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int x;
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int capability;
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struct mgcp_endpoint *p = sub->parent;
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capability = p->capability;
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if (codecs)
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capability = codecs;
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if (!strlen(sub->cxident) && rtp) {
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/* We don't have a CXident yet, store the destination and
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wait a bit */
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@ -1923,7 +1927,7 @@ static int transmit_modify_with_sdp(struct mgcp_subchannel *sub, struct ast_rtp
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}
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snprintf(local, sizeof(local), "p:20");
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for (x=1;x<= AST_FORMAT_MAX_AUDIO; x <<= 1) {
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if (p->capability & x) {
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if (capability & x) {
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snprintf(tmp, sizeof(tmp), ", a:%s", ast_rtp_lookup_mime_subtype(1, x));
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strcat(local, tmp);
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}
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@ -2298,7 +2302,7 @@ static void handle_response(struct mgcp_endpoint *p, struct mgcp_subchannel *sub
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}
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strncpy(sub->cxident, c, sizeof(sub->cxident) - 1);
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if (sub->tmpdest.sin_addr.s_addr) {
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transmit_modify_with_sdp(sub, NULL);
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transmit_modify_with_sdp(sub, NULL, 0);
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}
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}
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else {
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@ -3710,13 +3714,13 @@ static struct ast_rtp *mgcp_get_rtp_peer(struct ast_channel *chan)
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return NULL;
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}
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static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp)
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static int mgcp_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
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{
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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->pvt->pvt;
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if (sub) {
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transmit_modify_with_sdp(sub, rtp);
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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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return -1;
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@ -493,7 +493,7 @@ static int transmit_response_with_auth(struct sip_pvt *p, char *msg, struct sip_
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static int transmit_request(struct sip_pvt *p, char *msg, int inc, int reliable, int newbranch);
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static int transmit_request_with_auth(struct sip_pvt *p, char *msg, int inc, int reliable, int newbranch);
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static int transmit_invite(struct sip_pvt *p, char *msg, int sendsdp, char *auth, char *authheader, char *vxml_url,char *distinctive_ring, int init);
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static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp, struct ast_rtp *vrtp);
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static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codec);
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static int transmit_info_with_digit(struct sip_pvt *p, char digit);
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static int transmit_message_with_text(struct sip_pvt *p, char *text);
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static int transmit_refer(struct sip_pvt *p, char *dest);
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@ -3003,7 +3003,7 @@ static int add_digit(struct sip_request *req, char digit)
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}
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/*--- add_sdp: Add Session Description Protocol message ---*/
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static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *rtp, struct ast_rtp *vrtp)
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static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
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{
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int len;
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int codec;
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@ -3022,6 +3022,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
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char a[1024] = "";
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char a2[1024] = "";
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int x;
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int capability;
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struct sockaddr_in dest;
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struct sockaddr_in vdest = { 0, };
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/* XXX We break with the "recommendation" and send our IP, in order that our
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@ -3031,6 +3032,10 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
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ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
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return -1;
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}
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capability = p->jointcapability;
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if (codecs)
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capability = codecs & p->jointcapability;
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if (!p->sessionid) {
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p->sessionid = getpid();
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p->sessionversion = p->sessionid;
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@ -3073,7 +3078,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
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snprintf(t, sizeof(t), "t=0 0\r\n");
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snprintf(m, sizeof(m), "m=audio %d RTP/AVP", ntohs(dest.sin_port));
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snprintf(m2, sizeof(m2), "m=video %d RTP/AVP", ntohs(vdest.sin_port));
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if (p->jointcapability & p->prefcodec) {
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if (capability & p->prefcodec) {
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if (sip_debug_test_pvt(p))
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ast_verbose("Answering/Requesting with root capability %d\n", p->prefcodec);
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codec = ast_rtp_lookup_code(p->rtp, 1, p->prefcodec);
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@ -3094,7 +3099,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
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/* Start by sending our preferred codecs */
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cur = prefs;
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while(cur) {
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if ((p->jointcapability & cur->codec) && !(alreadysent & cur->codec)) {
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if ((capability & cur->codec) && !(alreadysent & cur->codec)) {
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if (sip_debug_test_pvt(p))
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ast_verbose("Answering with preferred capability 0x%x(%s)\n", cur->codec, ast_getformatname(cur->codec));
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codec = ast_rtp_lookup_code(p->rtp, 1, cur->codec);
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@ -3116,7 +3121,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
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}
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/* Now send any other common codecs, and non-codec formats: */
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for (x = 1; x <= (videosupport ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) {
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if ((p->jointcapability & x) && !(alreadysent & x)) {
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if ((capability & x) && !(alreadysent & x)) {
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if (sip_debug_test_pvt(p))
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ast_verbose("Answering with capability 0x%x(%s)\n", x, ast_getformatname(x));
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codec = ast_rtp_lookup_code(p->rtp, 1, x);
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@ -3161,7 +3166,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
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if ((sizeof(m) <= strlen(m) - 2) || (sizeof(m2) <= strlen(m2) - 2) || (sizeof(a) == strlen(a)) || (sizeof(a2) == strlen(a2)))
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ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
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len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m) + strlen(a);
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if ((p->vrtp) && (p->jointcapability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
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if ((p->vrtp) && (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
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len += strlen(m2) + strlen(a2);
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snprintf(costr, sizeof(costr), "%d", len);
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add_header(resp, "Content-Type", "application/sdp");
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@ -3173,7 +3178,7 @@ static int add_sdp(struct sip_request *resp, struct sip_pvt *p, struct ast_rtp *
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add_line(resp, t);
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add_line(resp, m);
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add_line(resp, a);
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if ((p->vrtp) && (p->jointcapability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */
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if ((p->vrtp) && (capability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */
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add_line(resp, m2);
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add_line(resp, a2);
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}
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@ -3205,7 +3210,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
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return -1;
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}
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respprep(&resp, p, msg, req);
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add_sdp(&resp, p, NULL, NULL);
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add_sdp(&resp, p, NULL, NULL, 0);
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return send_response(p, &resp, retrans, seqno);
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}
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@ -3272,7 +3277,7 @@ static int determine_firstline_parts( struct sip_request *req ) {
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/* transmit_reinvite_with_sdp: Transmit reinvite with SDP :-) ---*/
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/* A re-invite is basically a new INVITE with the same CALL-ID and TAG as the
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INVITE that opened the SIP dialogue */
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static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp, struct ast_rtp *vrtp)
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static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codec)
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{
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struct sip_request req;
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if (p->canreinvite == REINVITE_UPDATE)
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@ -3281,7 +3286,7 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp, st
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reqprep(&req, p, "INVITE", 0, 1);
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add_header(&req, "Allow", ALLOWED_METHODS);
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add_sdp(&req, p, rtp, vrtp);
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add_sdp(&req, p, rtp, vrtp, codec);
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/* Use this as the basis */
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copy_request(&p->initreq, &req);
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parse(&p->initreq);
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@ -3428,7 +3433,7 @@ static int transmit_invite(struct sip_pvt *p, char *cmd, int sdp, char *auth, ch
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}
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add_header(&req, "Allow", ALLOWED_METHODS);
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if (sdp) {
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add_sdp(&req, p, NULL, NULL);
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add_sdp(&req, p, NULL, NULL, 0);
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} else {
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add_header(&req, "Content-Length", "0");
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add_blank_header(&req);
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@ -7725,7 +7730,7 @@ static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan)
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return NULL;
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}
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static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp)
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static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codec)
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{
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struct sip_pvt *p;
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p = chan->pvt->pvt;
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@ -7739,7 +7744,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
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else
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memset(&p->vredirip, 0, sizeof(p->vredirip));
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if (!p->gotrefer) {
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transmit_reinvite_with_sdp(p, rtp, vrtp);
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transmit_reinvite_with_sdp(p, rtp, vrtp, codec);
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p->outgoing = 1;
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}
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return 0;
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@ -948,7 +948,7 @@ static struct ast_rtp *skinny_get_rtp_peer(struct ast_channel *chan)
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return NULL;
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}
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static int skinny_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp)
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static int skinny_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
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{
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struct skinny_subchannel *sub;
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sub = chan->pvt->pvt;
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@ -38,7 +38,7 @@ extern "C" {
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struct ast_rtp_protocol {
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struct ast_rtp *(*get_rtp_info)(struct ast_channel *chan); /* Get RTP struct, or NULL if unwilling to transfer */
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struct ast_rtp *(*get_vrtp_info)(struct ast_channel *chan); /* Get RTP struct, or NULL if unwilling to transfer */
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int (*set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer); /* Set RTP peer */
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int (*set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, int codecs); /* Set RTP peer */
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int (*get_codec)(struct ast_channel *chan);
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char *type;
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struct ast_rtp_protocol *next;
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41
rtp.c
41
rtp.c
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@ -1257,6 +1257,8 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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void *pvt0, *pvt1;
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int to;
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int codec0,codec1, oldcodec0, oldcodec1;
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memset(&vt0, 0, sizeof(vt0));
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memset(&vt1, 0, sizeof(vt1));
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memset(&vac0, 0, sizeof(vac0));
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@ -1307,10 +1309,15 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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ast_mutex_unlock(&c1->lock);
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return -2;
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}
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if (pr0->get_codec && pr1->get_codec) {
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int codec0,codec1;
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if (pr0->get_codec)
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codec0 = pr0->get_codec(c0);
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else
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codec0 = 0;
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if (pr1->get_codec)
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codec1 = pr1->get_codec(c1);
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else
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codec1 = 0;
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if (pr0->get_codec && pr1->get_codec) {
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/* Hey, we can't do reinvite if both parties speak diffrent codecs */
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if (!(codec0 & codec1)) {
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ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, cannot native bridge.\n",codec0,codec1);
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@ -1319,7 +1326,7 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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return -2;
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}
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}
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if (pr0->set_rtp_peer(c0, p1, vp1))
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if (pr0->set_rtp_peer(c0, p1, vp1, codec1))
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ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
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else {
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/* Store RTP peer */
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@ -1327,7 +1334,7 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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if (vp1)
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ast_rtp_get_peer(p1, &vac1);
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}
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if (pr1->set_rtp_peer(c1, p0, vp0))
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if (pr1->set_rtp_peer(c1, p0, vp0, codec0))
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ast_log(LOG_WARNING, "Channel '%s' failed to talk back to '%s'\n", c1->name, c0->name);
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else {
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/* Store RTP peer */
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@ -1340,17 +1347,19 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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cs[0] = c0;
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cs[1] = c1;
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cs[2] = NULL;
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oldcodec0 = codec0;
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oldcodec1 = codec1;
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for (;;) {
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if ((c0->pvt->pvt != pvt0) ||
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(c1->pvt->pvt != pvt1) ||
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(c0->masq || c0->masqr || c1->masq || c1->masqr)) {
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ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
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if (c0->pvt->pvt == pvt0) {
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if (pr0->set_rtp_peer(c0, NULL, NULL))
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if (pr0->set_rtp_peer(c0, NULL, NULL, 0))
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ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
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}
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if (c1->pvt->pvt == pvt1) {
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if (pr1->set_rtp_peer(c1, NULL, NULL))
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if (pr1->set_rtp_peer(c1, NULL, NULL, 0))
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ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
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}
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/* Tell it to try again later */
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@ -1359,23 +1368,29 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
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to = -1;
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ast_rtp_get_peer(p1, &t1);
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ast_rtp_get_peer(p0, &t0);
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if (pr0->get_codec)
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codec0 = pr0->get_codec(c0);
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if (pr1->get_codec)
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codec1 = pr1->get_codec(c1);
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if (vp1)
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ast_rtp_get_peer(vp1, &vt1);
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if (vp0)
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ast_rtp_get_peer(vp0, &vt0);
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if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1))) {
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ast_log(LOG_DEBUG, "Oooh, '%s' changed end address\n", c1->name);
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if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL))
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if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1)) || (codec1 != oldcodec1)) {
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ast_log(LOG_DEBUG, "Oooh, '%s' changed end address (format %d)\n", c1->name, codec1);
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if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, codec1))
|
||||
ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
|
||||
memcpy(&ac1, &t1, sizeof(ac1));
|
||||
memcpy(&vac1, &vt1, sizeof(vac1));
|
||||
oldcodec1 = codec1;
|
||||
}
|
||||
if (inaddrcmp(&t0, &ac0) || (vp0 && inaddrcmp(&vt0, &vac0))) {
|
||||
ast_log(LOG_DEBUG, "Oooh, '%s' changed end address\n", c0->name);
|
||||
if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL))
|
||||
ast_log(LOG_DEBUG, "Oooh, '%s' changed end address (format %d)\n", c0->name, codec0);
|
||||
if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, codec0))
|
||||
ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
|
||||
memcpy(&ac0, &t0, sizeof(ac0));
|
||||
memcpy(&vac0, &vt0, sizeof(vac0));
|
||||
oldcodec0 = codec0;
|
||||
}
|
||||
who = ast_waitfor_n(cs, 2, &to);
|
||||
if (!who) {
|
||||
|
|
@ -1393,11 +1408,11 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
|
|||
*rc = who;
|
||||
ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
|
||||
if ((c0->pvt->pvt == pvt0) && (!c0->_softhangup)) {
|
||||
if (pr0->set_rtp_peer(c0, NULL, NULL))
|
||||
if (pr0->set_rtp_peer(c0, NULL, NULL, 0))
|
||||
ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
|
||||
}
|
||||
if ((c1->pvt->pvt == pvt1) && (!c1->_softhangup)) {
|
||||
if (pr1->set_rtp_peer(c1, NULL, NULL))
|
||||
if (pr1->set_rtp_peer(c1, NULL, NULL, 0))
|
||||
ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
|
||||
}
|
||||
/* That's all we needed */
|
||||
|
|
|
|||
Loading…
Reference in New Issue