[asterisk-commits] chan sip.c: Set preferred rx payload type mapping on incomin... (asterisk[master])
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Aug 21 08:39:29 CDT 2015
Matt Jordan has submitted this change and it was merged.
Change subject: chan_sip.c: Set preferred rx payload type mapping on incoming offers.
......................................................................
chan_sip.c: Set preferred rx payload type mapping on incoming offers.
ASTERISK-25166
Reported by: Kevin Harwell
ASTERISK-17410
Reported by: Boris Fox
Change-Id: I7f04d5c8bee1126fee5fe6afbc39e45104469f4e
---
M channels/chan_sip.c
1 file changed, 20 insertions(+), 9 deletions(-)
Approvals:
Anonymous Coward #1000019: Verified
Matt Jordan: Looks good to me, approved
Ashley Sanders: Looks good to me, but someone else must approve
Joshua Colp: Looks good to me, but someone else must approve
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5a49392..1888091 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1205,7 +1205,7 @@
static const char *get_sdp_iterate(int* start, struct sip_request *req, const char *name);
static char get_sdp_line(int *start, int stop, struct sip_request *req, const char **value);
static int find_sdp(struct sip_request *req);
-static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action);
+static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action, int is_offer);
static int process_sdp_o(const char *o, struct sip_pvt *p);
static int process_sdp_c(const char *c, struct ast_sockaddr *addr);
static int process_sdp_a_sendonly(const char *a, int *sendonly);
@@ -9848,7 +9848,7 @@
Return 0 on success, a negative value on errors.
Must be called after find_sdp().
*/
-static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action)
+static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action, int is_offer)
{
int res = 0;
@@ -10537,6 +10537,16 @@
if (udptlportno == -1) {
change_t38_state(p, T38_DISABLED);
+ }
+
+ if (is_offer) {
+ /*
+ * Setup rx payload type mapping to prefer the mapping
+ * from the peer that the RFC says we SHOULD use.
+ */
+ ast_rtp_codecs_payloads_xover(&newaudiortp, &newaudiortp, NULL);
+ ast_rtp_codecs_payloads_xover(&newvideortp, &newvideortp, NULL);
+ ast_rtp_codecs_payloads_xover(&newtextrtp, &newtextrtp, NULL);
}
/* Now gather all of the codecs that we are asked for: */
@@ -23023,7 +23033,7 @@
if (p->invitestate != INV_CANCELLED) {
p->invitestate = INV_EARLY_MEDIA;
}
- res = process_sdp(p, req, SDP_T38_NONE);
+ res = process_sdp(p, req, SDP_T38_NONE, FALSE);
if (!req->ignore && p->owner) {
/* Queue a progress frame only if we have SDP in 180 or 182 */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
@@ -23108,7 +23118,7 @@
if (p->invitestate != INV_CANCELLED) {
p->invitestate = INV_EARLY_MEDIA;
}
- res = process_sdp(p, req, SDP_T38_NONE);
+ res = process_sdp(p, req, SDP_T38_NONE, FALSE);
if (!req->ignore && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
@@ -23134,7 +23144,8 @@
}
p->authtries = 0;
if (find_sdp(req)) {
- if ((res = process_sdp(p, req, SDP_T38_ACCEPT)) && !req->ignore) {
+ res = process_sdp(p, req, SDP_T38_ACCEPT, FALSE);
+ if (res && !req->ignore) {
if (!reinvite) {
/* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
/* For re-invites, we try to recover */
@@ -24414,7 +24425,7 @@
if (!req->ignore && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
if (find_sdp(req))
- process_sdp(p, req, SDP_T38_NONE);
+ process_sdp(p, req, SDP_T38_NONE, FALSE);
if (p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
@@ -25492,7 +25503,7 @@
}
/* Handle SDP here if we already have an owner */
if (find_sdp(req)) {
- if (process_sdp(p, req, SDP_T38_INITIATE)) {
+ if (process_sdp(p, req, SDP_T38_INITIATE, TRUE)) {
if (!ast_strlen_zero(sip_get_header(req, "Content-Encoding"))) {
/* Asterisk does not yet support any Content-Encoding methods. Always
* attempt to process the sdp, but return a 415 if a Content-Encoding header
@@ -25557,7 +25568,7 @@
/* We have a successful authentication, process the SDP portion if there is one */
if (find_sdp(req)) {
- if (process_sdp(p, req, SDP_T38_INITIATE)) {
+ if (process_sdp(p, req, SDP_T38_INITIATE, TRUE)) {
/* Asterisk does not yet support any Content-Encoding methods. Always
* attempt to process the sdp, but return a 415 if a Content-Encoding header
* was present after processing fails. */
@@ -28163,7 +28174,7 @@
p->pendinginvite = 0;
acked = __sip_ack(p, seqno, 1 /* response */, 0);
if (p->owner && find_sdp(req)) {
- if (process_sdp(p, req, SDP_T38_NONE)) {
+ if (process_sdp(p, req, SDP_T38_NONE, FALSE)) {
return -1;
}
if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
--
To view, visit https://gerrit.asterisk.org/1022
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Gerrit-MessageType: merged
Gerrit-Change-Id: I7f04d5c8bee1126fee5fe6afbc39e45104469f4e
Gerrit-PatchSet: 3
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Richard Mudgett <rmudgett at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Ashley Sanders <asanders at digium.com>
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>
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