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Merged revisions 338417 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/10

................
  r338417 | irroot | 2011-09-29 14:16:42 +0200 (Thu, 29 Sep 2011) | 19 lines
  
  Merged revisions 338416 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) | 12 lines
    
    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/trunk@338435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Gregory Nietsky
2011-09-29 12:22:43 +00:00
parent 383b073966
commit c4a7d0e2c7
2 changed files with 26 additions and 27 deletions
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51
channels/chan_sip.c
View File

@@ -5089,9 +5089,9 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) { if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
return -1; return -1;
} }
ast_rtp_instance_set_timeout(dialog->vrtp, global_rtptimeout); ast_rtp_instance_set_timeout(dialog->vrtp, dialog->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->vrtp, global_rtpholdtimeout); ast_rtp_instance_set_hold_timeout(dialog->vrtp, dialog->rtpholdtimeout);
ast_rtp_instance_set_keepalive(dialog->vrtp, global_rtpholdtimeout); ast_rtp_instance_set_keepalive(dialog->vrtp, dialog->rtpkeepalive);
ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1); ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
} }
@@ -5100,16 +5100,15 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
if (!(dialog->trtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) { if (!(dialog->trtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
return -1; return -1;
} }
ast_rtp_instance_set_timeout(dialog->trtp, global_rtptimeout); /* Do not timeout text as its not constant*/
ast_rtp_instance_set_hold_timeout(dialog->trtp, global_rtpholdtimeout); ast_rtp_instance_set_keepalive(dialog->trtp, dialog->rtpkeepalive);
ast_rtp_instance_set_keepalive(dialog->trtp, global_rtpholdtimeout);
ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1); 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_timeout(dialog->rtp, dialog->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->rtp, global_rtpholdtimeout); ast_rtp_instance_set_hold_timeout(dialog->rtp, dialog->rtpholdtimeout);
ast_rtp_instance_set_keepalive(dialog->rtp, global_rtpkeepalive); 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_RTCP, 1);
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, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
@@ -5170,6 +5169,9 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_string_field_set(dialog, engine, peer->engine); 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)) { if (dialog_initialize_rtp(dialog)) {
return -1; return -1;
} }
@@ -5177,23 +5179,10 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
if (dialog->rtp) { /* Audio */ 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, 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_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 */ /* Set Frame packetization */
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs); ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing; 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 */ /* XXX TODO: get fields directly from peer only as they are needed using dialog->relatedpeer */
ast_string_field_set(dialog, peername, peer->name); ast_string_field_set(dialog, peername, peer->name);
@@ -5221,7 +5210,6 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_copy_string(dialog->zone, peer->zone, sizeof(dialog->zone)); ast_copy_string(dialog->zone, peer->zone, sizeof(dialog->zone));
dialog->allowtransfer = peer->allowtransfer; dialog->allowtransfer = peer->allowtransfer;
dialog->jointnoncodeccapability = dialog->noncodeccapability; dialog->jointnoncodeccapability = dialog->noncodeccapability;
dialog->rtptimeout = peer->rtptimeout;
/* Update dialog authorization credentials */ /* Update dialog authorization credentials */
ao2_lock(peer); ao2_lock(peer);
@@ -5334,10 +5322,13 @@ static int create_addr(struct sip_pvt *dialog, const char *opeer, struct ast_soc
dialog->relatedpeer = sip_ref_peer(peer, "create_addr: setting dialog's relatedpeer pointer"); dialog->relatedpeer = sip_ref_peer(peer, "create_addr: setting dialog's relatedpeer pointer");
sip_unref_peer(peer, "create_addr: unref peer from sip_find_peer hashtab lookup"); sip_unref_peer(peer, "create_addr: unref peer from sip_find_peer hashtab lookup");
return res; return res;
} } else {
dialog->rtptimeout = global_rtptimeout;
if (dialog_initialize_rtp(dialog)) { dialog->rtpholdtimeout = global_rtpholdtimeout;
return -1; dialog->rtpkeepalive = global_rtpkeepalive;
if (dialog_initialize_rtp(dialog)) {
return -1;
}
} }
ast_string_field_set(dialog, tohost, hostport.host); ast_string_field_set(dialog, tohost, hostport.host);
@@ -15922,6 +15913,9 @@ static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
else else
p->noncodeccapability &= ~AST_RTP_DTMF; p->noncodeccapability &= ~AST_RTP_DTMF;
p->jointnoncodeccapability = p->noncodeccapability; p->jointnoncodeccapability = p->noncodeccapability;
p->rtptimeout = peer->rtptimeout;
p->rtpholdtimeout = peer->rtpholdtimeout;
p->rtpkeepalive = peer->rtpkeepalive;
if (!dialog_initialize_rtp(p)) { if (!dialog_initialize_rtp(p)) {
if (p->rtp) { if (p->rtp) {
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs); ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
@@ -16043,6 +16037,9 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
/* Finally, apply the guest policy */ /* Finally, apply the guest policy */
if (sip_cfg.allowguest) { if (sip_cfg.allowguest) {
get_rpid(p, req); get_rpid(p, req);
p->rtptimeout = global_rtptimeout;
p->rtpholdtimeout = global_rtpholdtimeout;
p->rtpkeepalive = global_rtpkeepalive;
if (!dialog_initialize_rtp(p)) { if (!dialog_initialize_rtp(p)) {
res = AUTH_SUCCESSFUL; res = AUTH_SUCCESSFUL;
} else { } else {

2
channels/sip/include/sip.h
View File

@@ -1072,6 +1072,8 @@ struct sip_pvt {
time_t lastrtprx; /*!< Last RTP received */ time_t lastrtprx; /*!< Last RTP received */
time_t lastrtptx; /*!< Last RTP sent */ time_t lastrtptx; /*!< Last RTP sent */
int rtptimeout; /*!< RTP timeout time */ 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_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 recv; /*!< Received as */
struct ast_sockaddr ourip; /*!< Our IP (as seen from the outside) */ struct ast_sockaddr ourip; /*!< Our IP (as seen from the outside) */