Commit 2184ab5493 for asterisk.org
commit 2184ab5493a8c4b5f075b2ba97522c9f9fe38313
Author: Mehrdad Seifzadeh <mehrdad.seifzadeh@gmail.com>
Date: Mon Jul 6 15:28:56 2026 +0330
res_pjsip: Use unique payload types for bundled streams
Ensure that locally generated media sections within the same BUNDLE
group use unique RTP payload types.
Reusing the same payload type across bundled media sections makes RTP
stream association ambiguous when payload-type-based demultiplexing is
used.
Fixes: #2010
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 1b0b820e2f..bb7bfbf978 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -441,6 +441,150 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
SCOPE_EXIT_RTN();
}
+
+/*! \brief Determine whether a payload type is already advertised in a BUNDLE group. */
+static int bundle_payload_type_used(struct ast_sip_session *session,
+ const pjmedia_sdp_session *sdp, const pjmedia_sdp_media *media,
+ int bundle_group, int payload)
+{
+ int media_index;
+
+ for (media_index = 0; media_index < sdp->media_count; ++media_index) {
+ struct ast_sip_session_media *other_session_media;
+ int format_index;
+
+ if (media_index >= AST_VECTOR_SIZE(&session->pending_media_state->sessions)) {
+ break;
+ }
+
+ other_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, media_index);
+ if (!other_session_media || other_session_media->bundle_group != bundle_group) {
+ continue;
+ }
+
+ for (format_index = 0; format_index < sdp->media[media_index]->desc.fmt_count; ++format_index) {
+ if (pj_strtoul(&sdp->media[media_index]->desc.fmt[format_index]) == payload) {
+ return 1;
+ }
+ }
+ }
+
+ /* The current media section is not in the SDP yet, so inspect it separately. */
+ for (media_index = 0; media_index < media->desc.fmt_count; ++media_index) {
+ if (pj_strtoul(&media->desc.fmt[media_index]) == payload) {
+ return 1;
+ }
+ }
+
+ return 0;
+}
+
+/*! \brief Check whether a payload slot is either free or already mapped to the requested format. */
+static int payload_type_compatible(struct ast_rtp_codecs *codecs, int payload,
+ struct ast_format *format)
+{
+ struct ast_rtp_payload_type *type;
+ int compatible;
+
+ type = ast_rtp_codecs_get_payload(codecs, payload);
+ if (!type) {
+ return 1;
+ }
+
+ compatible = type->asterisk_format
+ && ast_format_cmp(type->format, format) == AST_FORMAT_CMP_EQUAL;
+ ao2_ref(type, -1);
+
+ return compatible;
+}
+
+/*! \brief Ensure an RTP codecs structure can receive a format using a payload type. */
+static int ensure_payload_type_mapping(struct ast_rtp_codecs *codecs, int payload,
+ struct ast_format *format)
+{
+ struct ast_rtp_payload_type *type;
+ int matches;
+
+ type = ast_rtp_codecs_get_payload(codecs, payload);
+ if (type) {
+ matches = type->asterisk_format
+ && ast_format_cmp(type->format, format) == AST_FORMAT_CMP_EQUAL;
+ ao2_ref(type, -1);
+ return matches ? 0 : -1;
+ }
+
+ return ast_rtp_codecs_payload_set_rx(codecs, payload, format) < 0 ? -1 : 0;
+}
+
+/*! \brief Assign a payload type that is unique across the generated BUNDLE group. */
+static int assign_unique_bundle_payload(struct ast_sip_session *session,
+ struct ast_sip_session_media *session_media,
+ struct ast_sip_session_media *session_media_transport,
+ const pjmedia_sdp_session *sdp, const pjmedia_sdp_media *media,
+ struct ast_format *format)
+{
+ struct ast_rtp_codecs *transport_codecs =
+ ast_rtp_instance_get_codecs(session_media_transport->rtp);
+ struct ast_rtp_codecs *stream_codecs =
+ ast_rtp_instance_get_codecs(session_media->rtp);
+ int preferred;
+ int candidate;
+ int offset;
+
+ preferred = ast_rtp_codecs_payload_code(stream_codecs, 1, format, 0);
+ if (preferred < 0) {
+ return -1;
+ }
+
+ /* Preserve the stream's normal payload when it is not used by another m= section. */
+ if (!bundle_payload_type_used(session, sdp, media, session_media->bundle_group, preferred)
+ && payload_type_compatible(transport_codecs, preferred, format)
+ && payload_type_compatible(stream_codecs, preferred, format)) {
+ candidate = preferred;
+ goto configure;
+ }
+
+ /* Remap duplicate static or dynamic payloads into the dynamic range. */
+ candidate = preferred >= AST_RTP_PT_FIRST_DYNAMIC
+ ? preferred + 1 : AST_RTP_PT_FIRST_DYNAMIC;
+ for (offset = 0; offset < AST_RTP_MAX_PT - AST_RTP_PT_FIRST_DYNAMIC; ++offset) {
+ if (candidate >= AST_RTP_MAX_PT) {
+ candidate = AST_RTP_PT_FIRST_DYNAMIC;
+ }
+
+ if (!bundle_payload_type_used(session, sdp, media, session_media->bundle_group, candidate)
+ && payload_type_compatible(transport_codecs, candidate, format)
+ && payload_type_compatible(stream_codecs, candidate, format)) {
+ goto configure;
+ }
+
+ ++candidate;
+ }
+
+ ast_log(LOG_WARNING, "No unique dynamic RTP payload type available for bundled stream %s format %s\n",
+ S_OR(session_media->mid, "<unknown>"), ast_format_get_name(format));
+ return -1;
+
+configure:
+ if (ensure_payload_type_mapping(transport_codecs, candidate, format)
+ || ensure_payload_type_mapping(stream_codecs, candidate, format)) {
+ ast_log(LOG_WARNING, "Unable to configure RTP payload type %d for bundled stream %s format %s\n",
+ candidate, S_OR(session_media->mid, "<unknown>"), ast_format_get_name(format));
+ return -1;
+ }
+
+ ast_debug(2, "Assigned RTP payload type %d to bundled stream %s format %s\n",
+ candidate, S_OR(session_media->mid, "<unknown>"), ast_format_get_name(format));
+ return candidate;
+}
+
+/*! \brief Determine whether the SDP currently being generated is a local offer. */
+static int generating_local_offer(const struct ast_sip_session *session)
+{
+ return !session->inv_session->neg
+ || pjmedia_sdp_neg_get_state(session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE;
+}
+
static int apply_cap_to_bundled(struct ast_sip_session_media *session_media,
struct ast_sip_session_media *session_media_transport,
struct ast_stream *asterisk_stream, struct ast_format_cap *joint)
@@ -2020,13 +2164,24 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
* conflicts.
*/
if (session_media_transport != session_media) {
- if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media_transport->rtp), 1, format, 0)) == -1) {
+ if (generating_local_offer(session)) {
+ rtp_code = assign_unique_bundle_payload(session, session_media,
+ session_media_transport, sdp, media, format);
+ } else {
+ rtp_code = ast_rtp_codecs_payload_code(
+ ast_rtp_instance_get_codecs(session_media_transport->rtp), 1, format, 0);
+ if (rtp_code != -1) {
+ /* Our instance has to match the offered payload number. */
+ ast_rtp_codecs_payload_set_rx(
+ ast_rtp_instance_get_codecs(session_media->rtp), rtp_code, format);
+ }
+ }
+
+ if (rtp_code == -1) {
ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format));
ao2_ref(format, -1);
continue;
}
- /* Our instance has to match the payload number though */
- ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media->rtp), rtp_code, format);
} else {
if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0)) == -1) {
ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format));