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mark RTP sessions that are not carrying DTMF

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allow native bridging of RTP sessions that are not carrying DTMF even when the bridge needs to listen to DTMF (when SIP INFO is used for DTMF, for example)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@27559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
2006-05-16 22:11:02 +00:00
parent ca8f9083c0
commit c3dce57d95
3 changed files with 35 additions and 5 deletions
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7
channels/chan_sip.c
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@@ -2195,6 +2195,7 @@ static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer)
ast_rtp_destroy(r->vrtp);
r->vrtp = NULL;
}
ast_rtp_setdtmf(r->rtp, ast_test_flag(&r->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
r->prefs = peer->prefs;
natflags = ast_test_flag(&r->flags[0], SIP_NAT) & SIP_NAT_ROUTE;
if (r->rtp) {
@@ -3524,9 +3525,12 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
free(p);
return NULL;
}
ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
ast_rtp_settos(p->rtp, global_tos_audio);
if (p->vrtp)
if (p->vrtp) {
ast_rtp_settos(p->vrtp, global_tos_video);
ast_rtp_setdtmf(p->vrtp, 0);
}
p->rtptimeout = global_rtptimeout;
p->rtpholdtimeout = global_rtpholdtimeout;
p->rtpkeepalive = global_rtpkeepalive;
@@ -11170,6 +11174,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
get_rdnis(p, NULL); /* Get redirect information */
extract_uri(p, req); /* Get the Contact URI */
build_contact(p); /* Build our contact header */
ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
if (gotdest) {
if (gotdest == 1 && ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {