Merged revisions 323370 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines

  Add rtpkeepalives back to 1.8

  The RTP-engine conversion left out support for handling rtpkeepalives.
  This patch adds them back.

  (closes issue ASTERISK-17304)
  Reported by: lmadsen

  Review: https://reviewboard.asterisk.org/r/1226/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323374 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 92b2f3d..7940b01 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5061,6 +5061,7 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
                }
                ast_rtp_instance_set_timeout(dialog->vrtp, global_rtptimeout);
                ast_rtp_instance_set_hold_timeout(dialog->vrtp, global_rtpholdtimeout);
+               ast_rtp_instance_set_keepalive(dialog->vrtp, global_rtpholdtimeout);
 
                ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
        }
@@ -5071,12 +5072,14 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
                }
                ast_rtp_instance_set_timeout(dialog->trtp, global_rtptimeout);
                ast_rtp_instance_set_hold_timeout(dialog->trtp, global_rtpholdtimeout);
+               ast_rtp_instance_set_keepalive(dialog->trtp, global_rtpholdtimeout);
 
                ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
        }
 
        ast_rtp_instance_set_timeout(dialog->rtp, global_rtptimeout);
        ast_rtp_instance_set_hold_timeout(dialog->rtp, global_rtpholdtimeout);
+       ast_rtp_instance_set_keepalive(dialog->rtp, global_rtpkeepalive);
 
        ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, 1);
        ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
@@ -5144,6 +5147,7 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
                ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
                ast_rtp_instance_set_timeout(dialog->rtp, peer->rtptimeout);
                ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
+               ast_rtp_instance_set_keepalive(dialog->rtp, peer->rtpkeepalive);
                /* Set Frame packetization */
                ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
                dialog->autoframing = peer->autoframing;
@@ -5151,10 +5155,12 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
        if (dialog->vrtp) { /* Video */
                ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
                ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
+               ast_rtp_instance_set_keepalive(dialog->vrtp, peer->rtpkeepalive);
        }
        if (dialog->trtp) { /* Realtime text */
                ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
                ast_rtp_instance_set_hold_timeout(dialog->trtp, peer->rtpholdtimeout);
+               ast_rtp_instance_set_keepalive(dialog->trtp, peer->rtpkeepalive);
        }
 
        /* XXX TODO: get fields directly from peer only as they are needed using dialog->relatedpeer */
@@ -25396,11 +25402,19 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
        }
 
        /* If we have no timers set, return now */
-       if (!ast_rtp_instance_get_timeout(dialog->rtp) && !ast_rtp_instance_get_hold_timeout(dialog->rtp)) {
+       if (!ast_rtp_instance_get_keepalive(dialog->rtp) && !ast_rtp_instance_get_timeout(dialog->rtp) && !ast_rtp_instance_get_hold_timeout(dialog->rtp)) {
                dialog_unlink_rtpcheck(dialog);
                return;
        }
 
+       /* Check AUDIO RTP keepalives */
+       if (dialog->lastrtptx && ast_rtp_instance_get_keepalive(dialog->rtp) &&
+                   (t > dialog->lastrtptx + ast_rtp_instance_get_keepalive(dialog->rtp))) {
+               /* Need to send an empty RTP packet */
+               dialog->lastrtptx = time(NULL);
+               ast_rtp_instance_sendcng(dialog->rtp, 0);
+       }
+
        /*! \todo Check video RTP keepalives
 
                Do we need to move the lastrtptx to the RTP structure to have one for audio and one

diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index f8bf749..e04303c 100644 (file)
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -377,6 +377,8 @@ struct ast_rtp_engine {
        void (*stun_request)(struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username);
        /*! Callback to get the transcodeable formats supported. result returned in ast_format_cap *result */
        void (*available_formats)(struct ast_rtp_instance *instance, struct ast_format_cap *to_endpoint, struct ast_format_cap *to_asterisk, struct ast_format_cap *result);
+       /*! Callback to send CNG */
+       int (*sendcng)(struct ast_rtp_instance *instance, int level);
        /*! Linked list information */
        AST_RWLIST_ENTRY(ast_rtp_engine) entry;
 };
@@ -1713,6 +1715,24 @@ void ast_rtp_instance_set_timeout(struct ast_rtp_instance *instance, int timeout
 void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int timeout);
 
 /*!
+ * \brief Set the RTP keepalive interval
+ *
+ * \param instance The RTP instance
+ * \param period Value to set the keepalive interval to
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_instance_set_keepalive(instance, 5000);
+ * \endcode
+ *
+ * This sets the RTP keepalive interval on 'instance' to be 5000.
+ *
+ * \since 1.8
+ */
+void ast_rtp_instance_set_keepalive(struct ast_rtp_instance *instance, int timeout);
+
+/*!
  * \brief Get the RTP timeout value
  *
  * \param instance The RTP instance
@@ -1751,6 +1771,25 @@ int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance);
 int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance);
 
 /*!
+ * \brief Get the RTP keepalive interval
+ *
+ * \param instance The RTP instance
+ *
+ * \retval period Keepalive interval value
+ *
+ * Example usage:
+ *
+ * \code
+ * int interval = ast_rtp_instance_get_keepalive(instance);
+ * \endcode
+ *
+ * This gets the RTP keepalive interval value for the RTP instance pointed to by 'instance'.
+ *
+ * \since 1.8
+ */
+int ast_rtp_instance_get_keepalive(struct ast_rtp_instance *instance);
+
+/*!
  * \brief Get the RTP engine in use on an RTP instance
  *
  * \param instance The RTP instance
@@ -1809,6 +1848,17 @@ struct ast_rtp_glue *ast_rtp_instance_get_active_glue(struct ast_rtp_instance *i
  */
 struct ast_channel *ast_rtp_instance_get_chan(struct ast_rtp_instance *instance);
 
+/*!
+ * \brief Send a comfort noise packet to the RTP instance
+ *
+ * \param instance The RTP instance
+ * \param level Magnitude of the noise level
+ *
+ * \retval 0 Success
+ * \retval non-zero Failure
+ */
+int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level);
+
 int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *policy);
 struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance);
 

diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 3c5ef8b..a1c4605 100644 (file)
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -66,6 +66,8 @@ struct ast_rtp_instance {
        int timeout;
        /*! RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
        int holdtimeout;
+       /*! RTP keepalive interval */
+       int keepalive;
        /*! DTMF mode in use */
        enum ast_rtp_dtmf_mode dtmf_mode;
        /*! Glue currently in use */
@@ -1781,6 +1783,11 @@ void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int ti
        instance->holdtimeout = timeout;
 }
 
+void ast_rtp_instance_set_keepalive(struct ast_rtp_instance *instance, int interval)
+{
+       instance->keepalive = interval;
+}
+
 int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance)
 {
        return instance->timeout;
@@ -1791,6 +1798,11 @@ int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance)
        return instance->holdtimeout;
 }
 
+int ast_rtp_instance_get_keepalive(struct ast_rtp_instance *instance)
+{
+       return instance->keepalive;
+}
+
 struct ast_rtp_engine *ast_rtp_instance_get_engine(struct ast_rtp_instance *instance)
 {
        return instance->engine;
@@ -1850,6 +1862,15 @@ struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance)
        return instance->srtp;
 }
 
+int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level)
+{
+       if (instance->engine->sendcng) {
+               return instance->engine->sendcng(instance, level);
+       }
+
+       return -1;
+}
+
 static void set_next_mime_type(const struct ast_format *format, int rtp_code, char *type, char *subtype, unsigned int sample_rate)
 {
        int x = mime_types_len;

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index dd71342..60f7eda 100644 (file)
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -272,6 +272,7 @@ static int ast_rtp_dtmf_compatible(struct ast_channel *chan0, struct ast_rtp_ins
 static void ast_rtp_stun_request(struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username);
 static void ast_rtp_stop(struct ast_rtp_instance *instance);
 static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos, const char* desc);
+static int ast_rtp_sendcng(struct ast_rtp_instance *instance, int level);
 
 /* RTP Engine Declaration */
 static struct ast_rtp_engine asterisk_rtp_engine = {
@@ -297,6 +298,7 @@ static struct ast_rtp_engine asterisk_rtp_engine = {
        .stun_request = ast_rtp_stun_request,
        .stop = ast_rtp_stop,
        .qos = ast_rtp_qos_set,
+       .sendcng = ast_rtp_sendcng,
 };
 
 static inline int rtp_debug_test_addr(struct ast_sockaddr *addr)
@@ -2591,6 +2593,49 @@ static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos,
        return ast_set_qos(rtp->s, tos, cos, desc);
 }
 
+/*! \brief generate comfort noice (CNG) */
+static int ast_rtp_sendcng(struct ast_rtp_instance *instance, int level)
+{
+       unsigned int *rtpheader;
+       int hdrlen = 12;
+       int res;
+       int payload;
+       char data[256];
+       struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+       struct ast_sockaddr remote_address = { {0,} };
+
+       ast_rtp_instance_get_remote_address(instance, &remote_address);
+
+       if (ast_sockaddr_isnull(&remote_address)) {
+               return -1;
+       }
+
+       payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_CN);
+
+       level = 127 - (level & 0x7f);
+       
+       rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
+
+       /* Get a pointer to the header */
+       rtpheader = (unsigned int *)data;
+       rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
+       rtpheader[1] = htonl(rtp->lastts);
+       rtpheader[2] = htonl(rtp->ssrc); 
+       data[12] = level;
+
+       res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 1, 0, &remote_address);
+
+       if (res < 0) {
+               ast_log(LOG_ERROR, "RTP Comfort Noise Transmission error to %s: %s\n", ast_sockaddr_stringify(&remote_address), strerror(errno));
+       } else if (rtp_debug_test_addr(&remote_address)) {
+               ast_verbose("Sent Comfort Noise RTP packet to %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+                               ast_sockaddr_stringify(&remote_address),
+                               AST_RTP_CN, rtp->seqno, rtp->lastdigitts, res - hdrlen);
+       }
+
+       return res;
+}
+
 static char *rtp_do_debug_ip(struct ast_cli_args *a)
 {
        char *arg = ast_strdupa(a->argv[4]);