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Add new SIP cli command "sip show channelstats" that displays some QoS data (if we have RTCP reports
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
This commit is contained in:
Olle Johansson
+113
-11
@@ -311,6 +311,22 @@ enum invitestates {
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INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */
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};
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/*! \brief Readable descriptions of device states.
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\note Should be aligned to above table as index */
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static const struct invstate2stringtable {
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const enum invitestates state;
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const char const *desc;
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} invitestate2string[] = {
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{INV_NONE, "None" },
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{INV_CALLING, "Calling (Trying)"},
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{INV_PROCEEDING, "Proceeding "},
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{INV_EARLY_MEDIA, "Early media"},
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{INV_COMPLETED, "Completed (done)"},
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{INV_CONFIRMED, "Confirmed (up)"},
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{INV_TERMINATED, "Done"},
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{INV_CANCELLED, "Cancelled"}
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};
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enum xmittype {
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XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
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If it fails, it's critical and will cause a teardown of the session */
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@@ -453,6 +469,14 @@ struct sip_proxy {
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/* Room for a SRV record chain based on the name */
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};
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/*! \brief argument for the 'show channels|subscriptions' callback. */
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struct __show_chan_arg {
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int fd;
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int subscriptions;
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int numchans; /* return value */
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};
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/*! \brief States whether a SIP message can create a dialog in Asterisk. */
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enum can_create_dialog {
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CAN_NOT_CREATE_DIALOG,
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@@ -2018,6 +2042,7 @@ static char *complete_sip_user(const char *word, int state, int flags2);
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static char *complete_sip_show_user(const char *line, const char *word, int pos, int state);
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static char *complete_sipnotify(const char *line, const char *word, int pos, int state);
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static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
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static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
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static char *sip_show_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
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static char *sip_do_debug_ip(int fd, char *arg);
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static char *sip_do_debug_peer(int fd, char *arg);
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@@ -14117,6 +14142,91 @@ static char *sip_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_arg
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return CLI_SUCCESS;
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}
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/*! \brief Callback for show_chanstats */
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static int show_chanstats_cb(void *__cur, void *__arg, int flags)
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{
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#define FORMAT2 "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s (%-2.2s) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"
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#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"
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struct sip_pvt *cur = __cur;
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unsigned int rxcount;
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unsigned int txcount;
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char durbuf[10];
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int duration;
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int durh, durm, durs;
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struct ast_channel *c = cur->owner;
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struct __show_chan_arg *arg = __arg;
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int fd = arg->fd;
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if (cur->subscribed != NONE) /* Subscriptions */
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return 0; /* don't care, we scan all channels */
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if (!cur->rtp) {
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if (sipdebug)
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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");
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return 0; /* don't care, we scan all channels */
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}
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rxcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXCOUNT);
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txcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXCOUNT);
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/* Find the duration of this channel */
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if (c && c->cdr && !ast_tvzero(c->cdr->start)) {
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duration = (int)(ast_tvdiff_ms(ast_tvnow(), c->cdr->start) / 1000);
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durh = duration / 3600;
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durm = (duration % 3600) / 60;
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durs = duration % 60;
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snprintf(durbuf, sizeof(durbuf), "%02d:%02d:%02d", durh, durm, durs);
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} else {
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durbuf[0] = '\0';
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}
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/* Print stats for every call with RTP */
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ast_cli(fd, FORMAT,
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ast_inet_ntoa(cur->sa.sin_addr),
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cur->callid,
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durbuf,
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rxcount > (unsigned int) 100000 ? (unsigned int) (rxcount)/(unsigned int) 1000 : rxcount,
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rxcount > (unsigned int) 100000 ? "K":" ",
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ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS),
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rxcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) ? (unsigned int) (ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) / rxcount * 100) : 0,
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ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXJITTER),
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txcount > (unsigned int) 100000 ? (unsigned int) (txcount)/(unsigned int) 1000 : txcount,
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txcount > (unsigned int) 100000 ? "K":" ",
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ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS),
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txcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS) ? (unsigned int) (ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS)/ txcount * 100) : 0,
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ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXJITTER)
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);
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arg->numchans++;
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return 0; /* don't care, we scan all channels */
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}
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/*! \brief SIP show channelstats CLI (main function) */
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static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
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{
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struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };
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switch (cmd) {
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case CLI_INIT:
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e->command = "sip show channelstats";
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e->usage =
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"Usage: sip show channelstats\n"
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" Lists all currently active SIP channel's RTCP statistics.\n"
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" Note that calls in the much optimized RTP P2P bridge mode will not show any packets here.";
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return NULL;
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}
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if (a->argc != 3)
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return CLI_SHOWUSAGE;
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ast_cli(a->fd, FORMAT2, "Peer", "Call ID", "Duration", "Recv: Pack", "Lost", "%", "Jitter", "Send: Pack", "Lost", "Jitter");
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/* iterate on the container and invoke the callback on each item */
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ao2_t_callback(dialogs, OBJ_NODATA, show_chanstats_cb, &arg, "callback to sip show chanstats");
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ast_cli(a->fd, "%d active SIP channel%s\n", arg.numchans, (arg.numchans != 1) ? "s" : "");
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return CLI_SUCCESS;
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}
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#undef FORMAT
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#undef FORMAT2
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/*! \brief List global settings for the SIP channel */
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static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
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{
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@@ -14334,13 +14444,6 @@ static const struct cfsubscription_types *find_subscription_type(enum subscripti
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* that support iteration through callbacks will be a lot easier.
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*/
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/*! \brief argument for the 'show channels|subscriptions' callback. */
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struct __show_chan_arg {
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int fd;
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int subscriptions;
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int numchans; /* return value */
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};
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#define FORMAT4 "%-15.15s %-10.10s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6d\n"
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#define FORMAT3 "%-15.15s %-10.10s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6s\n"
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#define FORMAT2 "%-15.15s %-10.10s %-15.15s %-15.15s %-7.7s %-15.15s %-6.6s\n"
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@@ -14436,8 +14539,6 @@ static char *sip_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_
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* given position. As many functions of this kind, each invokation has
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* O(state) time complexity so be careful in using it.
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*/
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static char *complete_sipch(const char *line, const char *word, int pos, int state)
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{
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int which=0;
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@@ -23001,12 +23102,13 @@ static int reload(void)
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static struct ast_cli_entry cli_sip_do_history_deprecated = AST_CLI_DEFINE(sip_do_history_deprecated, "Enable/Disable SIP history");
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/*! \brief SIP Cli commands definition */
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static struct ast_cli_entry cli_sip[] = {
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AST_CLI_DEFINE(sip_show_channels, "List active SIP channels/subscriptions"),
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AST_CLI_DEFINE(sip_show_channels, "List active SIP channels or subscriptions"),
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AST_CLI_DEFINE(sip_show_channelstats, "List statistics for active SIP channels"),
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AST_CLI_DEFINE(sip_show_domains, "List our local SIP domains."),
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AST_CLI_DEFINE(sip_show_inuse, "List all inuse/limits"),
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AST_CLI_DEFINE(sip_show_objects, "List all SIP object allocations"),
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AST_CLI_DEFINE(sip_show_peers, "List defined SIP peers"),
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AST_CLI_DEFINE(sip_dbdump, "dump peer info into realtime db sql format"),
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AST_CLI_DEFINE(sip_dbdump, "Dump peer info into realtime database SQL format"),
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AST_CLI_DEFINE(sip_show_registry, "List SIP registration status"),
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AST_CLI_DEFINE(sip_unregister, "Unregister (force expiration) a SIP peer from the registery\n"),
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AST_CLI_DEFINE(sip_show_settings, "Show SIP global settings"),
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@@ -69,6 +69,17 @@ enum ast_rtp_get_result {
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AST_RTP_TRY_NATIVE,
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};
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/*! \brief Variables used in ast_rtcp_get function */
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enum ast_rtp_qos_vars {
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AST_RTP_TXCOUNT,
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AST_RTP_RXCOUNT,
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AST_RTP_TXJITTER,
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AST_RTP_RXJITTER,
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AST_RTP_RXPLOSS,
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AST_RTP_TXPLOSS,
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AST_RTP_RTT
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};
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struct ast_rtp;
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/*! T.140 Redundancy structure*/
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struct rtp_red;
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@@ -268,6 +279,10 @@ int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1);
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/*! \brief Get QOS stats on a RTP channel */
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int ast_rtp_get_qos(struct ast_rtp *rtp, const char *qos, char *buf, unsigned int buflen);
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/*! \brief Return RTP and RTCP QoS values */
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unsigned int ast_rtp_get_qosvalue(struct ast_rtp *rtp, enum ast_rtp_qos_vars value);
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/*! \brief Set RTPAUDIOQOS(...) variables on a channel when it is being hung up */
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void ast_rtp_set_vars(struct ast_channel *chan, struct ast_rtp *rtp);
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+31
@@ -2626,6 +2626,37 @@ void ast_rtp_reset(struct ast_rtp *rtp)
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rtp->rxseqno = 0;
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}
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/*! Get QoS values from RTP and RTCP data (used in "sip show channelstats") */
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unsigned int ast_rtp_get_qosvalue(struct ast_rtp *rtp, enum ast_rtp_qos_vars value)
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{
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if (rtp == NULL) {
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if (option_debug > 1)
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ast_log(LOG_DEBUG, "NO RTP Structure? Kidding me? \n");
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return 0;
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}
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if (option_debug > 1 && rtp->rtcp == NULL) {
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ast_log(LOG_DEBUG, "NO RTCP structure. Maybe in RTP p2p bridging mode? \n");
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}
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switch (value) {
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case AST_RTP_TXCOUNT:
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return (unsigned int) rtp->txcount;
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case AST_RTP_RXCOUNT:
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return (unsigned int) rtp->rxcount;
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case AST_RTP_TXJITTER:
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return (unsigned int) (rtp->rxjitter * 100.0);
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case AST_RTP_RXJITTER:
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return (unsigned int) rtp->rtcp ? (rtp->rtcp->reported_jitter / (unsigned int) 65536.0) : 0;
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case AST_RTP_RXPLOSS:
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return rtp->rtcp ? (rtp->rtcp->expected_prior - rtp->rtcp->received_prior) : 0;
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case AST_RTP_TXPLOSS:
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return rtp->rtcp ? rtp->rtcp->reported_lost : 0;
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case AST_RTP_RTT:
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return (unsigned int) rtp->rtcp ? rtp->rtcp->rtt * 100 : 0;
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}
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return 0; /* To make the compiler happy */
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}
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static double __ast_rtp_get_qos(struct ast_rtp *rtp, const char *qos, int *found)
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{
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*found = 1;
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