Add new SIP cli command "sip show channelstats" that displays some QoS data (if we...

and not use the p2p rtp bridge). I could not find a way to detect us using the p2p bridge, which
would be nice.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128197 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 280e128e3684b255ef8c04c4624ee3f6feaf0c2d..7ff5eed29079c519d0c4c1950876b95af8042abe 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -311,6 +311,22 @@ enum invitestates {
        INV_CANCELLED = 7,      /*!< Transaction cancelled by client or server in non-terminated state */
 };
 
+/*! \brief Readable descriptions of device states.
+       \note Should be aligned to above table as index */
+static const struct invstate2stringtable {
+       const enum invitestates state;
+       const char const *desc;
+} invitestate2string[] = {
+       {INV_NONE,              "None"  },
+       {INV_CALLING,           "Calling (Trying)"},
+       {INV_PROCEEDING,        "Proceeding "},
+       {INV_EARLY_MEDIA,       "Early media"},
+       {INV_COMPLETED,         "Completed (done)"},
+       {INV_CONFIRMED,         "Confirmed (up)"},
+       {INV_TERMINATED,        "Done"},
+       {INV_CANCELLED,         "Cancelled"}
+};
+
 enum xmittype {
        XMIT_CRITICAL = 2,              /*!< Transmit critical SIP message reliably, with re-transmits.
                                               If it fails, it's critical and will cause a teardown of the session */
@@ -453,6 +469,14 @@ struct sip_proxy {
        /* Room for a SRV record chain based on the name */
 };
 
+/*! \brief argument for the 'show channels|subscriptions' callback. */
+struct __show_chan_arg { 
+       int fd;
+       int subscriptions;
+       int numchans;   /* return value */
+};
+
+
 /*! \brief States whether a SIP message can create a dialog in Asterisk. */
 enum can_create_dialog {
        CAN_NOT_CREATE_DIALOG,
@@ -2018,6 +2042,7 @@ static char *complete_sip_user(const char *word, int state, int flags2);
 static char *complete_sip_show_user(const char *line, const char *word, int pos, int state);
 static char *complete_sipnotify(const char *line, const char *word, int pos, int state);
 static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
+static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
 static char *sip_show_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
 static char *sip_do_debug_ip(int fd, char *arg);
 static char *sip_do_debug_peer(int fd, char *arg);
@@ -14117,6 +14142,91 @@ static char *sip_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_arg
        return CLI_SUCCESS;
 }
 
+/*! \brief Callback for show_chanstats */
+static int show_chanstats_cb(void *__cur, void *__arg, int flags)
+{
+#define FORMAT2 "%-15.15s  %-11.11s  %-8.8s %-10.10s  %-10.10s (%-2.2s) %-6.6s %-10.10s  %-10.10s ( %%) %-6.6s\n"
+#define FORMAT  "%-15.15s  %-11.11s  %-8.8s %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u\n"
+       struct sip_pvt *cur = __cur;
+       unsigned int rxcount;
+       unsigned int txcount;
+       char durbuf[10];
+        int duration;
+        int durh, durm, durs;
+       struct ast_channel *c = cur->owner;
+       struct __show_chan_arg *arg = __arg;
+       int fd = arg->fd;
+
+
+       if (cur->subscribed != NONE) /* Subscriptions */
+               return 0;       /* don't care, we scan all channels */
+
+       if (!cur->rtp) {
+               if (sipdebug)
+                       ast_cli(fd, "%-15.15s  %-11.11s (inv state: %s) -- %s\n", ast_inet_ntoa(cur->sa.sin_addr), cur->callid, invitestate2string[cur->invitestate].desc, "-- No RTP active");
+               return 0;       /* don't care, we scan all channels */
+       }
+       rxcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXCOUNT);
+       txcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXCOUNT);
+
+       /* Find the duration of this channel */
+       if (c && c->cdr && !ast_tvzero(c->cdr->start)) {
+               duration = (int)(ast_tvdiff_ms(ast_tvnow(), c->cdr->start) / 1000);
+               durh = duration / 3600;
+               durm = (duration % 3600) / 60;
+               durs = duration % 60;
+               snprintf(durbuf, sizeof(durbuf), "%02d:%02d:%02d", durh, durm, durs);
+       } else {
+               durbuf[0] = '\0';
+       }
+       /* Print stats for every call with RTP */
+       ast_cli(fd, FORMAT, 
+               ast_inet_ntoa(cur->sa.sin_addr), 
+               cur->callid, 
+               durbuf,
+               rxcount > (unsigned int) 100000 ? (unsigned int) (rxcount)/(unsigned int) 1000 : rxcount,
+               rxcount > (unsigned int) 100000 ? "K":" ",
+               ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS),
+               rxcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) ? (unsigned int) (ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) / rxcount * 100) : 0,
+               ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXJITTER),
+               txcount > (unsigned int) 100000 ? (unsigned int) (txcount)/(unsigned int) 1000 : txcount,
+               txcount > (unsigned int) 100000 ? "K":" ",
+               ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS),
+               txcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS) ? (unsigned int) (ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS)/ txcount * 100) : 0,
+               ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXJITTER)
+       );
+       arg->numchans++;
+
+       return 0;       /* don't care, we scan all channels */
+}
+
+/*! \brief SIP show channelstats CLI (main function) */
+static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
+{
+       struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };
+
+       switch (cmd) {
+       case CLI_INIT:
+               e->command = "sip show channelstats";
+               e->usage =
+                       "Usage: sip show channelstats\n"
+                       "       Lists all currently active SIP channel's RTCP statistics.\n"
+                       "       Note that calls in the much optimized RTP P2P bridge mode will not show any packets here.";
+               return NULL;
+       }
+
+       if (a->argc != 3)
+               return CLI_SHOWUSAGE;
+
+       ast_cli(a->fd, FORMAT2, "Peer", "Call ID", "Duration", "Recv: Pack", "Lost", "%", "Jitter", "Send: Pack", "Lost", "Jitter");
+       /* iterate on the container and invoke the callback on each item */
+       ao2_t_callback(dialogs, OBJ_NODATA, show_chanstats_cb, &arg, "callback to sip show chanstats");
+       ast_cli(a->fd, "%d active SIP channel%s\n", arg.numchans, (arg.numchans != 1) ? "s" : ""); 
+       return CLI_SUCCESS;
+}
+#undef FORMAT
+#undef FORMAT2
+
 /*! \brief List global settings for the SIP channel */
 static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
 {
@@ -14334,13 +14444,6 @@ static const struct cfsubscription_types *find_subscription_type(enum subscripti
  * that support iteration through callbacks will be a lot easier.
  */
 
-/*! \brief argument for the 'show channels|subscriptions' callback. */
-struct __show_chan_arg { 
-       int fd;
-       int subscriptions;
-       int numchans;   /* return value */
-};
-
 #define FORMAT4 "%-15.15s  %-10.10s  %-15.15s  %-15.15s  %-13.13s  %-15.15s %-10.10s %-6.6d\n"
 #define FORMAT3 "%-15.15s  %-10.10s  %-15.15s  %-15.15s  %-13.13s  %-15.15s %-10.10s %-6.6s\n"
 #define FORMAT2 "%-15.15s  %-10.10s  %-15.15s  %-15.15s  %-7.7s  %-15.15s %-6.6s\n"
@@ -14436,8 +14539,6 @@ static char *sip_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_
  * given position. As many functions of this kind, each invokation has
  * O(state) time complexity so be careful in using it.
  */
-
-
 static char *complete_sipch(const char *line, const char *word, int pos, int state)
 {
        int which=0;
@@ -23001,12 +23102,13 @@ static int reload(void)
 static struct ast_cli_entry cli_sip_do_history_deprecated = AST_CLI_DEFINE(sip_do_history_deprecated, "Enable/Disable SIP history");
 /*! \brief SIP Cli commands definition */
 static struct ast_cli_entry cli_sip[] = {
-       AST_CLI_DEFINE(sip_show_channels, "List active SIP channels/subscriptions"),
+       AST_CLI_DEFINE(sip_show_channels, "List active SIP channels or subscriptions"),
+       AST_CLI_DEFINE(sip_show_channelstats, "List statistics for active SIP channels"),
        AST_CLI_DEFINE(sip_show_domains, "List our local SIP domains."),
        AST_CLI_DEFINE(sip_show_inuse, "List all inuse/limits"),
        AST_CLI_DEFINE(sip_show_objects, "List all SIP object allocations"),
        AST_CLI_DEFINE(sip_show_peers, "List defined SIP peers"),
-       AST_CLI_DEFINE(sip_dbdump, "dump peer info into realtime db sql format"),
+       AST_CLI_DEFINE(sip_dbdump, "Dump peer info into realtime database SQL format"),
        AST_CLI_DEFINE(sip_show_registry, "List SIP registration status"),
        AST_CLI_DEFINE(sip_unregister, "Unregister (force expiration) a SIP peer from the registery\n"),
        AST_CLI_DEFINE(sip_show_settings, "Show SIP global settings"),

diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 38fe8639ceb5f34eea93808290bc848c2fb87815..800519572a1e1a60f49332f56c24c188e4717b1b 100644 (file)
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -69,6 +69,17 @@ enum ast_rtp_get_result {
        AST_RTP_TRY_NATIVE,
 };
 
+/*! \brief Variables used in ast_rtcp_get function */
+enum ast_rtp_qos_vars {
+       AST_RTP_TXCOUNT,
+       AST_RTP_RXCOUNT,
+       AST_RTP_TXJITTER,
+       AST_RTP_RXJITTER,
+       AST_RTP_RXPLOSS,
+       AST_RTP_TXPLOSS,
+       AST_RTP_RTT
+};
+
 struct ast_rtp;
 /*! T.140 Redundancy structure*/
 struct rtp_red;
@@ -268,6 +279,10 @@ int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1);
 
 /*! \brief Get QOS stats on a RTP channel */
 int ast_rtp_get_qos(struct ast_rtp *rtp, const char *qos, char *buf, unsigned int buflen);
+
+/*! \brief Return RTP and RTCP QoS values */
+unsigned int ast_rtp_get_qosvalue(struct ast_rtp *rtp, enum ast_rtp_qos_vars value);
+
 /*! \brief Set RTPAUDIOQOS(...) variables on a channel when it is being hung up */
 void ast_rtp_set_vars(struct ast_channel *chan, struct ast_rtp *rtp);
 

diff --git a/main/rtp.c b/main/rtp.c
index 146e8c09363810f55a8fbc65223362e84d288cb8..4a8bf2b7bef64fb76b0e7bf9b3130cf70293eecf 100644 (file)
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -2626,6 +2626,37 @@ void ast_rtp_reset(struct ast_rtp *rtp)
        rtp->rxseqno = 0;
 }
 
+/*! Get QoS values from RTP and RTCP data (used in "sip show channelstats") */
+unsigned int ast_rtp_get_qosvalue(struct ast_rtp *rtp, enum ast_rtp_qos_vars value)
+{
+       if (rtp == NULL) {
+               if (option_debug > 1)
+                       ast_log(LOG_DEBUG, "NO RTP Structure? Kidding me? \n");
+               return 0;
+       }
+       if (option_debug > 1 && rtp->rtcp == NULL) {
+               ast_log(LOG_DEBUG, "NO RTCP structure. Maybe in RTP p2p bridging mode? \n");
+       }
+
+       switch (value) {
+       case AST_RTP_TXCOUNT:
+               return (unsigned int) rtp->txcount;
+       case AST_RTP_RXCOUNT:
+               return (unsigned int) rtp->rxcount;
+       case AST_RTP_TXJITTER:
+               return (unsigned int) (rtp->rxjitter * 100.0);
+       case AST_RTP_RXJITTER:
+               return (unsigned int) rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int) 65536.0) : 0;
+       case AST_RTP_RXPLOSS:
+               return rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0;
+       case AST_RTP_TXPLOSS:
+               return rtp->rtcp ? rtp->rtcp->reported_lost : 0;
+       case AST_RTP_RTT:
+               return (unsigned int) rtp->rtcp ? rtp->rtcp->rtt * 100 : 0;
+       }
+       return 0;       /* To make the compiler happy */
+}
+
 static double __ast_rtp_get_qos(struct ast_rtp *rtp, const char *qos, int *found)
 {
        *found = 1;