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

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 5a49392..1888091 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1205,7 +1205,7 @@ static void try_suggested_sip_codec(struct sip_pvt *p);
 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 @@ static int has_media_stream(struct sip_pvt *p, enum media_type m)
        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;
 
@@ -10539,6 +10539,16 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
                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: */
        ast_rtp_codecs_payload_formats(&newaudiortp, peercapability, &peernoncodeccapability);
        ast_rtp_codecs_payload_formats(&newvideortp, vpeercapability, &vpeernoncodeccapability);
@@ -23023,7 +23033,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                        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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                        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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
                }
                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 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
                                        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 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
                        }
                        /* 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 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
 
                /* 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 @@ static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct as
                        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)) {