diff options
Diffstat (limited to 'channels/chan_sip.c')
| -rw-r--r-- | channels/chan_sip.c | 69 |
1 files changed, 55 insertions, 14 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 61a981abe..737927918 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -5063,15 +5063,25 @@ static int sip_hangup(struct ast_channel *ast) } } else { /* Call is in UP state, send BYE */ if (!p->pendinginvite) { + struct ast_channel *bridge = ast_bridged_channel(oldowner); char *audioqos = ""; char *videoqos = ""; char *textqos = ""; + if (p->rtp) - audioqos = ast_rtp_get_quality(p->rtp, NULL); + ast_rtp_set_vars(oldowner, p->rtp); + + if (bridge) { + struct sip_pvt *q = bridge->tech_pvt; + + if (IS_SIP_TECH(bridge->tech) && q->rtp) + ast_rtp_set_vars(bridge, q->rtp); + } + if (p->vrtp) - videoqos = ast_rtp_get_quality(p->vrtp, NULL); + videoqos = ast_rtp_get_quality(p->vrtp, NULL, RTPQOS_SUMMARY); if (p->trtp) - textqos = ast_rtp_get_quality(p->trtp, NULL); + textqos = ast_rtp_get_quality(p->trtp, NULL, RTPQOS_SUMMARY); /* Send a hangup */ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1); @@ -18429,10 +18439,13 @@ static int acf_channel_read(struct ast_channel *chan, const char *funcname, char ast_rtp_get_peer(p->vrtp, &sin); else if (!strcasecmp(args.type, "text")) ast_rtp_get_peer(p->trtp, &sin); + else + return -1; snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port)); } else if (!strcasecmp(args.param, "rtpqos")) { struct ast_rtp_quality qos; + struct ast_rtp *rtp = p->rtp; memset(&qos, 0, sizeof(qos)); @@ -18442,11 +18455,13 @@ static int acf_channel_read(struct ast_channel *chan, const char *funcname, char args.field = "all"; if (strcasecmp(args.type, "AUDIO") == 0) { - all = ast_rtp_get_quality(p->rtp, &qos); + all = ast_rtp_get_quality(rtp = p->rtp, &qos, RTPQOS_SUMMARY); } else if (strcasecmp(args.type, "VIDEO") == 0) { - all = ast_rtp_get_quality(p->vrtp, &qos); + all = ast_rtp_get_quality(rtp = p->vrtp, &qos, RTPQOS_SUMMARY); } else if (strcasecmp(args.type, "TEXT") == 0) { - all = ast_rtp_get_quality(p->trtp, &qos); + all = ast_rtp_get_quality(rtp = p->trtp, &qos, RTPQOS_SUMMARY); + } else { + return -1; } if (strcasecmp(args.field, "local_ssrc") == 0) @@ -18469,6 +18484,8 @@ static int acf_channel_read(struct ast_channel *chan, const char *funcname, char snprintf(buf, buflen, "%.0f", qos.rtt * 1000.0); else if (strcasecmp(args.field, "all") == 0) ast_copy_string(buf, all, buflen); + else if (!ast_rtp_get_qos(rtp, args.field, buf, buflen)) + ; else { ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname); return -1; @@ -18501,23 +18518,47 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req) /* Get RTCP quality before end of call */ if (p->do_history || p->owner) { - char *audioqos, *videoqos, *textqos; - if (p->rtp) { - audioqos = ast_rtp_get_quality(p->rtp, NULL); - if (p->do_history) + struct ast_channel *bridge = ast_bridged_channel(p->owner); + char *videoqos, *textqos; + + if (p->rtp) { + if (p->do_history) { + char *audioqos, + *audioqos_jitter, + *audioqos_loss, + *audioqos_rtt; + + audioqos = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_SUMMARY); + audioqos_jitter = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_JITTER); + audioqos_loss = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_LOSS); + audioqos_rtt = ast_rtp_get_quality(p->rtp, NULL, RTPQOS_RTT); + append_history(p, "RTCPaudio", "Quality:%s", audioqos); - if (p->owner) - pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos); + append_history(p, "RTCPaudioJitter", "Quality:%s", audioqos_jitter); + append_history(p, "RTCPaudioLoss", "Quality:%s", audioqos_loss); + append_history(p, "RTCPaudioRTT", "Quality:%s", audioqos_rtt); + } + + ast_rtp_set_vars(p->owner, p->rtp); } + + if (bridge) { + struct sip_pvt *q = bridge->tech_pvt; + + if (IS_SIP_TECH(bridge->tech) && q->rtp) + ast_rtp_set_vars(bridge, q->rtp); + } + if (p->vrtp) { - videoqos = ast_rtp_get_quality(p->vrtp, NULL); + videoqos = ast_rtp_get_quality(p->vrtp, NULL, RTPQOS_SUMMARY); if (p->do_history) append_history(p, "RTCPvideo", "Quality:%s", videoqos); if (p->owner) pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos); } + if (p->trtp) { - textqos = ast_rtp_get_quality(p->trtp, NULL); + textqos = ast_rtp_get_quality(p->trtp, NULL, RTPQOS_SUMMARY); if (p->do_history) append_history(p, "RTCPtext", "Quality:%s", textqos); if (p->owner) |