Merge "chan_sip.c: Fix T.38 issues caused by leaving a bridge."

This commit is contained in:
Joshua Colp 2016-03-01 06:00:43 -06:00 committed by Gerrit Code Review
commit d0b26c3133
1 changed files with 30 additions and 9 deletions

View File

@ -7522,8 +7522,10 @@ static int interpret_t38_parameters(struct sip_pvt *p, const struct ast_control_
AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
change_t38_state(p, T38_REJECTED);
transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
} else if (p->t38.state == T38_ENABLED)
} else if (p->t38.state == T38_ENABLED) {
change_t38_state(p, T38_DISABLED);
transmit_reinvite_with_sdp(p, FALSE, FALSE);
}
break;
case AST_T38_REQUEST_PARMS: { /* Application wants remote's parameters re-sent */
struct ast_control_t38_parameters parameters = p->t38.their_parms;
@ -10677,6 +10679,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
} else if (udptlportno > 0) {
if (debug)
ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n");
/* Force media to go through us for T.38. */
memset(&p->redirip, 0, sizeof(p->redirip));
/* Prevent audio RTCP reads */
if (p->owner) {
ast_channel_set_fd(p->owner, 1, -1);
@ -13788,6 +13794,29 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int old
{
struct sip_request req;
if (t38version) {
/* Force media to go through us for T.38. */
memset(&p->redirip, 0, sizeof(p->redirip));
}
if (p->rtp) {
if (t38version) {
/* Silence RTCP while audio RTP is inactive */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
if (p->owner) {
/* Prevent audio RTCP reads */
ast_channel_set_fd(p->owner, 1, -1);
}
} else if (ast_sockaddr_isnull(&p->redirip)) {
/* Enable RTCP since it will be inactive if we're coming back
* with this reinvite */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
if (p->owner) {
/* Enable audio RTCP reads */
ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
}
}
}
reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);
add_header(&req, "Allow", ALLOWED_METHODS);
@ -32780,14 +32809,6 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
} else if (!ast_sockaddr_isnull(&p->redirip)) {
memset(&p->redirip, 0, sizeof(p->redirip));
changed = 1;
if (p->rtp) {
/* Enable RTCP since it will be inactive if we're coming back
* from a reinvite */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
/* Enable audio RTCP reads */
ast_channel_set_fd(chan, 1, ast_rtp_instance_fd(p->rtp, 1));
}
}
if (vinstance) {