diff options
author | Luigi Rizzo <rizzo@icir.org> | 2006-10-07 11:28:38 +0000 |
---|---|---|
committer | Luigi Rizzo <rizzo@icir.org> | 2006-10-07 11:28:38 +0000 |
commit | bc1e5f77aff06e933db10167e49a4092b5d723cc (patch) | |
tree | 7696d6e8cb32bf1a0a15fe475cfe0d2ab01810ba | |
parent | b37dc86c743ded57cc022aaf843d018906337f38 (diff) |
put common code in a function to avoid repetitions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-rw-r--r-- | channels/chan_sip.c | 86 |
1 files changed, 35 insertions, 51 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 629cf7798..0f4a1e50e 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -11766,6 +11766,18 @@ static int handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_requ return 1; } +/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */ +static void stop_data_flows(struct sip_pvt *p) +{ + /* Immediately stop RTP, VRTP and UDPTL as applicable */ + if (p->rtp) + ast_rtp_stop(p->rtp); + if (p->vrtp) + ast_rtp_stop(p->vrtp); + if (p->udptl) + ast_udptl_stop(p->udptl); +} + /*! \brief Handle SIP response in dialogue */ /* XXX only called by handle_request */ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno) @@ -11955,18 +11967,9 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_ if ((option_verbose > 2) && (resp != 487)) ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr)); ast_set_flag(&p->flags[0], SIP_ALREADYGONE); - if (p->rtp) { - /* Immediately stop RTP */ - ast_rtp_stop(p->rtp); - } - if (p->vrtp) { - /* Immediately stop VRTP */ - ast_rtp_stop(p->vrtp); - } - if (p->udptl) { - /* Immediately stop UDPTL */ - ast_udptl_stop(p->udptl); - } + + stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */ + /* XXX Locking issues?? XXX */ switch(resp) { case 300: /* Multiple Choices */ @@ -13696,18 +13699,8 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req) ast_log(LOG_DEBUG, "Got CANCEL on an answered call. Ignoring... \n"); return 0; } - if (p->rtp) { - /* Immediately stop RTP */ - ast_rtp_stop(p->rtp); - } - if (p->vrtp) { - /* Immediately stop VRTP */ - ast_rtp_stop(p->vrtp); - } - if (p->udptl) { - /* Immediately stop UDPTL */ - ast_udptl_stop(p->udptl); - } + stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */ + if (p->owner) ast_queue_hangup(p->owner); else @@ -13728,7 +13721,6 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req) struct ast_channel *c=NULL; int res; struct ast_channel *bridged_to; - char *audioqos = NULL, *videoqos = NULL; if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE)) transmit_response_reliable(p, "487 Request Terminated", &p->initreq); @@ -13737,35 +13729,27 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req) check_via(p, req); ast_set_flag(&p->flags[0], SIP_ALREADYGONE); - if (p->rtp) - audioqos = ast_rtp_get_quality(p->rtp); - if (p->vrtp) - videoqos = ast_rtp_get_quality(p->vrtp); - /* Get RTCP quality before end of call */ - if (recordhistory) { - if (p->rtp) - append_history(p, "RTCPaudio", "Quality:%s", audioqos); - if (p->vrtp) - append_history(p, "RTCPvideo", "Quality:%s", videoqos); + if (recordhistory || p->owner) { + char *audioqos, *videoqos; + if (p->rtp) { + audioqos = ast_rtp_get_quality(p->rtp); + if (recordhistory) + append_history(p, "RTCPaudio", "Quality:%s", audioqos); + if (p->owner) + pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos); + } + if (p->vrtp) { + videoqos = ast_rtp_get_quality(p->vrtp); + if (recordhistory) + append_history(p, "RTCPvideo", "Quality:%s", videoqos); + if (p->owner) + pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos); + } } - if (p->rtp) { - if (p->owner) - pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos); - /* Immediately stop RTP */ - ast_rtp_stop(p->rtp); - } - if (p->vrtp) { - if (p->owner) - pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos); - /* Immediately stop VRTP */ - ast_rtp_stop(p->vrtp); - } - if (p->udptl) { - /* Immediately stop UDPTL */ - ast_udptl_stop(p->udptl); - } + stop_data_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */ + if (!ast_strlen_zero(get_header(req, "Also"))) { ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead\n", ast_inet_ntoa(p->recv.sin_addr)); |