From: Gregory Nietsky Date: Thu, 29 Sep 2011 12:13:05 +0000 (+0000) Subject: The rtptimeout setting is ignored on a per peer basis. X-Git-Tag: 1.8.8.0-rc1~24 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=e7d6d7ee19d9dd3a0113106c03f0f263b2c7d41a;p=thirdparty%2Fasterisk.git The rtptimeout setting is ignored on a per peer basis. Not only is the rtptimeout ignored in some cases but rtpkeepalive and rtpholdtimeout is affected. this commit also removes rtptimeout/rtpholdtimeout on text rtp. (closes issue ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338416 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 68a88754fe..e7263d6455 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -5047,9 +5047,9 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog) if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) { return -1; } - 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_timeout(dialog->vrtp, dialog->rtptimeout); + ast_rtp_instance_set_hold_timeout(dialog->vrtp, dialog->rtpholdtimeout); + ast_rtp_instance_set_keepalive(dialog->vrtp, dialog->rtpkeepalive); ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1); } @@ -5058,16 +5058,15 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog) if (!(dialog->trtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) { return -1; } - 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); + /* Do not timeout text as its not constant*/ + ast_rtp_instance_set_keepalive(dialog->trtp, dialog->rtpkeepalive); 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_timeout(dialog->rtp, dialog->rtptimeout); + ast_rtp_instance_set_hold_timeout(dialog->rtp, dialog->rtpholdtimeout); + ast_rtp_instance_set_keepalive(dialog->rtp, dialog->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); @@ -5128,6 +5127,9 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer) ast_string_field_set(dialog, engine, peer->engine); + dialog->rtptimeout = peer->rtptimeout; + dialog->rtpholdtimeout = peer->rtpholdtimeout; + dialog->rtpkeepalive = peer->rtpkeepalive; if (dialog_initialize_rtp(dialog)) { return -1; } @@ -5135,23 +5137,10 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer) if (dialog->rtp) { /* Audio */ ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833); 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; } - 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 */ ast_string_field_set(dialog, peername, peer->name); @@ -5178,7 +5167,6 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer) dialog->pickupgroup = peer->pickupgroup; dialog->allowtransfer = peer->allowtransfer; dialog->jointnoncodeccapability = dialog->noncodeccapability; - dialog->rtptimeout = peer->rtptimeout; /* Update dialog authorization credentials */ ao2_lock(peer); @@ -5291,10 +5279,13 @@ static int create_addr(struct sip_pvt *dialog, const char *opeer, struct ast_soc dialog->relatedpeer = ref_peer(peer, "create_addr: setting dialog's relatedpeer pointer"); unref_peer(peer, "create_addr: unref peer from find_peer hashtab lookup"); return res; - } - - if (dialog_initialize_rtp(dialog)) { - return -1; + } else { + dialog->rtptimeout = global_rtptimeout; + dialog->rtpholdtimeout = global_rtpholdtimeout; + dialog->rtpkeepalive = global_rtpkeepalive; + if (dialog_initialize_rtp(dialog)) { + return -1; + } } ast_string_field_set(dialog, tohost, hostport.host); @@ -15569,6 +15560,9 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of, else p->noncodeccapability &= ~AST_RTP_DTMF; p->jointnoncodeccapability = p->noncodeccapability; + p->rtptimeout = peer->rtptimeout; + p->rtpholdtimeout = peer->rtpholdtimeout; + p->rtpkeepalive = peer->rtpkeepalive; if (!dialog_initialize_rtp(p)) { if (p->rtp) { ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs); @@ -15690,6 +15684,9 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ /* Finally, apply the guest policy */ if (sip_cfg.allowguest) { get_rpid(p, req); + p->rtptimeout = global_rtptimeout; + p->rtpholdtimeout = global_rtpholdtimeout; + p->rtpkeepalive = global_rtpkeepalive; if (!dialog_initialize_rtp(p)) { res = AUTH_SUCCESSFUL; } else { diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h index 02da6814ef..ffd4cd2a31 100644 --- a/channels/sip/include/sip.h +++ b/channels/sip/include/sip.h @@ -1038,6 +1038,8 @@ struct sip_pvt { time_t lastrtprx; /*!< Last RTP received */ time_t lastrtptx; /*!< Last RTP sent */ int rtptimeout; /*!< RTP timeout time */ + int rtpholdtimeout; /*!< RTP timeout time on hold*/ + int rtpkeepalive; /*!< RTP send packets for keepalive */ struct ast_ha *directmediaha; /*!< Which IPs are allowed to interchange direct media with this peer - copied from sip_peer */ struct ast_sockaddr recv; /*!< Received as */ struct ast_sockaddr ourip; /*!< Our IP (as seen from the outside) */