Merged revisions 338417 via svnmerge from

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

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 083f88e..05af4f3 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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))) {
                        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);
        }
@@ -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))) {
                        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);
@@ -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);
 
+       dialog->rtptimeout = peer->rtptimeout;
+       dialog->rtpholdtimeout = peer->rtpholdtimeout;
+       dialog->rtpkeepalive = peer->rtpkeepalive;
        if (dialog_initialize_rtp(dialog)) {
                return -1;
        }
@@ -5177,23 +5179,10 @@ 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);
@@ -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));
        dialog->allowtransfer = peer->allowtransfer;
        dialog->jointnoncodeccapability = dialog->noncodeccapability;
-       dialog->rtptimeout = peer->rtptimeout;
 
        /* Update dialog authorization credentials */
        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");
                sip_unref_peer(peer, "create_addr: unref peer from sip_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);
@@ -15922,6 +15913,9 @@ 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);
@@ -16043,6 +16037,9 @@ 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 baec6ee..fc75ff4 100644 (file)
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -1072,6 +1072,8 @@ 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) */