res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits

After change made in 624f509 to add support for non 8K RFC 4733/2833 digits,
Asterisk would only accept RFC 4733/2833 offers that matched the sample rate of
the negotiated codec(s).

This change allows Asterisk to accept 8K RFC 4733/2833 offers if the UAC
offfers 8K RFC 4733/2833 but negotiates for a non 8K bitrate codec.

A number of corresponding tests in tests/channels/pjsip/dtmf_sdp also needed to
be re-written to allow for these scenarios.

Fixes: #776
(cherry picked from commit 7d53986262268b9b34641457dcc1cc9437265759)

diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index cb555b77977dfe9dc50c40f20228c56c2e23e27e..b3519f9176c17949a6495556acd9349328ff0220 100644 (file)
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -106,6 +106,9 @@ extern "C" {
  */
 #define MAX_CHANNEL_ID 152
 
+/*!< DTMF samples per second */
+#define DEFAULT_DTMF_SAMPLE_RATE_MS    8000
+
 struct ast_rtp_instance;
 struct ast_rtp_glue;
 

diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 5e992350ae6dc73ef4f69e4ee2ff33c6b3b38808..c86a5059140e80bf61b09710297979a904fc123d 100644 (file)
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -1989,10 +1989,8 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
                }
 
                if ((attr = generate_rtpmap_attr(session, media, pool, rtp_code, 1, format, 0))) {
-                       int newrate = ast_rtp_lookup_sample_rate2(1, format, 0);
                        int i, added = 0;
-                       media->attr[media->attr_count++] = attr;
-
+                       int newrate = ast_rtp_lookup_sample_rate2(1, format, 0);
                        if (build_dtmf_sample_rates) {
                                for (i = 0; i < AST_VECTOR_SIZE(&sample_rates); i++) {
                                        /* Only add if we haven't already processed this sample rate. For instance
@@ -2007,6 +2005,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
                                        AST_VECTOR_APPEND(&sample_rates, newrate);
                                }
                        }
+                       media->attr[media->attr_count++] = attr;
                }
 
                if ((attr = generate_fmtp_attr(pool, format, rtp_code))) {
@@ -2032,7 +2031,6 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
                                continue;
                        }
 
-
                        if (index != AST_RTP_DTMF) {
                                rtp_code = ast_rtp_codecs_payload_code(
                                                                ast_rtp_instance_get_codecs(session_media->rtp), 0, NULL, index);
@@ -2046,7 +2044,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
                                 * Walk through the possible bitrates for the RFC 2833/4733 digits and generate the rtpmap
                                 * attributes.
                                 */
-                               int i;
+                               int i, found_default_offer = 0;
                                for (i = 0; i < AST_VECTOR_SIZE(&sample_rates); i++) {
                                        rtp_code = ast_rtp_codecs_payload_code_sample_rate(
                                                                        ast_rtp_instance_get_codecs(session_media->rtp), 0, NULL, index, AST_VECTOR_GET(&sample_rates, i));
@@ -2055,12 +2053,31 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
                                                continue;
                                        }
 
+                                       if (AST_VECTOR_GET(&sample_rates, i) == DEFAULT_DTMF_SAMPLE_RATE_MS) {
+                                               /* we found and added a default offer, so no need to include a default one.*/
+                                               found_default_offer = 1;
+                                       }
+
                                        if ((attr = generate_rtpmap_attr2(session, media, pool, rtp_code, 0, NULL, index, AST_VECTOR_GET(&sample_rates, i)))) {
                                                media->attr[media->attr_count++] = attr;
                                                snprintf(tmp, sizeof(tmp), "%d 0-16", (rtp_code));
                                                attr = pjmedia_sdp_attr_create(pool, "fmtp", pj_cstr(&stmp, tmp));
                                                media->attr[media->attr_count++] = attr;
+                                       }
+                               }
+
+                               /* If we weren't able to add any matching RFC 2833/4733, assume this endpoint is using a
+                                * mismatched 8K offer and try to add one as a fall-back/default.
+                                */
+                               if (!found_default_offer) {
+                                       rtp_code = ast_rtp_codecs_payload_code_sample_rate(
+                                                                       ast_rtp_instance_get_codecs(session_media->rtp), 0, NULL, index, DEFAULT_DTMF_SAMPLE_RATE_MS);
 
+                                       if (rtp_code != -1 && (attr = generate_rtpmap_attr2(session, media, pool, rtp_code, 0, NULL, index, DEFAULT_DTMF_SAMPLE_RATE_MS))) {
+                                               media->attr[media->attr_count++] = attr;
+                                               snprintf(tmp, sizeof(tmp), "%d 0-16", (rtp_code));
+                                               attr = pjmedia_sdp_attr_create(pool, "fmtp", pj_cstr(&stmp, tmp));
+                                               media->attr[media->attr_count++] = attr;
                                        }
                                }
                        }

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index db763e3ed02a757e5f90a818e1f5305e9758a9f3..c3314612eb78ded9efe542d30d353fcd7865a6a0 100644 (file)
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -4326,11 +4326,17 @@ static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit)
        /* Grab the matching DTMF type payload */
        payload = ast_rtp_codecs_payload_code_tx_sample_rate(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_DTMF, sample_rate);
 
-       /* If this returns -1, we are being asked to send digits for a sample rate that is outside
-          what was negotiated for. Fall back if possible. */
+       /* If this returns -1, we are using a codec with a sample rate that does not have a matching RFC 2833/4733
+          offer. The offer may have included a default-rate one that doesn't match the codec rate, so try to use that. */
+       if (payload == -1) {
+               sample_rate = DEFAULT_DTMF_SAMPLE_RATE_MS;
+               payload = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_DTMF);
+       }
+       /* No default-rate offer either, trying to send a digit outside of what was negotiated for. */
        if (payload == -1) {
                return -1;
        }
+
        ast_test_suite_event_notify("DTMF_BEGIN", "Digit: %d\r\nPayload: %d\r\nRate: %d\r\n", digit, payload, sample_rate);
        ast_debug(1, "Sending digit '%d' at rate %d with payload %d\n", digit, sample_rate, payload);