mark RTP sessions that are not carrying DTMF

allow native bridging of RTP sessions that are not carrying DTMF even when the bridge needs to listen to DTMF (when SIP INFO is used for DTMF, for example)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27559 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d623df5ee1aa2e4df95ceb1465912b5598bbcaab..189e4f0646c4b835960ad1c49a6acc739a806fb3 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2195,6 +2195,7 @@ static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer)
                ast_rtp_destroy(r->vrtp);
                r->vrtp = NULL;
        }
+       ast_rtp_setdtmf(r->rtp, ast_test_flag(&r->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
        r->prefs = peer->prefs;
        natflags = ast_test_flag(&r->flags[0], SIP_NAT) & SIP_NAT_ROUTE;
        if (r->rtp) {
@@ -3524,9 +3525,12 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
                        free(p);
                        return NULL;
                }
+               ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
                ast_rtp_settos(p->rtp, global_tos_audio);
-               if (p->vrtp)
+               if (p->vrtp) {
                        ast_rtp_settos(p->vrtp, global_tos_video);
+                       ast_rtp_setdtmf(p->vrtp, 0);
+               }
                p->rtptimeout = global_rtptimeout;
                p->rtpholdtimeout = global_rtpholdtimeout;
                p->rtpkeepalive = global_rtpkeepalive;
@@ -11170,6 +11174,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
                get_rdnis(p, NULL);                     /* Get redirect information */
                extract_uri(p, req);                    /* Get the Contact URI */
                build_contact(p);                       /* Build our contact header */
+               ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
 
                if (gotdest) {
                        if (gotdest == 1 && ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {

diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 7235639f16d4303fe10b9eb2603eec96d7906d77..5b03d1d861dfe613d1ae70385391b95d69b90ea4 100644 (file)
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -151,6 +151,9 @@ char *ast_rtp_lookup_mime_multiple(char *buf, int size, const int capability, co
 
 void ast_rtp_setnat(struct ast_rtp *rtp, int nat);
 
+/*! \brief Indicate whether this RTP session is carrying DTMF or not */
+void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf);
+
 int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
 
 int ast_rtp_proto_register(struct ast_rtp_protocol *proto);

diff --git a/rtp.c b/rtp.c
index 841bbbd950dbadc8ed72834bfaf3a9758a03d03f..bfb2262038c09df6d221a27b8f1b72ab7163e05c 100644 (file)
--- a/rtp.c
+++ b/rtp.c
@@ -86,6 +86,7 @@ struct rtpPayloadType {
 #define FLAG_NAT_ACTIVE                        (3 << 1)
 #define FLAG_NAT_INACTIVE              (0 << 1)
 #define FLAG_NAT_INACTIVE_NOWARN       (1 << 1)
+#define FLAG_HAS_DTMF                  (1 << 3)
 
 /*! \brief RTP session description */
 struct ast_rtp {
@@ -434,6 +435,11 @@ void ast_rtp_setnat(struct ast_rtp *rtp, int nat)
        rtp->nat = nat;
 }
 
+void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf)
+{
+       ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF);
+}
+
 static struct ast_frame *send_dtmf(struct ast_rtp *rtp)
 {
        char iabuf[INET_ADDRSTRLEN];
@@ -1344,6 +1350,7 @@ struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io
        rtp->s = rtp_socket();
        rtp->ssrc = ast_random();
        rtp->seqno = ast_random() & 0xffff;
+       ast_set_flag(rtp, FLAG_HAS_DTMF);
        if (rtp->s < 0) {
                free(rtp);
                ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno));
@@ -1921,10 +1928,6 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
        memset(&vac0, 0, sizeof(vac0));
        memset(&vac1, 0, sizeof(vac1));
 
-       /* if need DTMF, cant native bridge */
-       if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1))
-               return AST_BRIDGE_FAILED_NOWARN;
-
        /* Lock channels */
        ast_channel_lock(c0);
        while(ast_channel_trylock(c1)) {
@@ -1966,6 +1969,25 @@ enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel
                ast_channel_unlock(c1);
                return AST_BRIDGE_FAILED_NOWARN;
        }
+
+       if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
+               /* can't bridge, we are carrying DTMF for this channel and the bridge
+                  needs it
+               */
+               ast_channel_unlock(c0);
+               ast_channel_unlock(c1);
+               return AST_BRIDGE_FAILED_NOWARN;
+       }
+
+       if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
+               /* can't bridge, we are carrying DTMF for this channel and the bridge
+                  needs it
+               */
+               ast_channel_unlock(c0);
+               ast_channel_unlock(c1);
+               return AST_BRIDGE_FAILED_NOWARN;
+       }
+
        /* Get codecs from both sides */
        codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
        codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;