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Diffstat (limited to 'res/res_pjsip_sdp_rtp.c')
-rw-r--r--res/res_pjsip_sdp_rtp.c16
1 files changed, 15 insertions, 1 deletions
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 75dc83929..a82475774 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -196,6 +196,20 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
if (session->endpoint->media.bind_rtp_to_media_address && !ast_strlen_zero(session->endpoint->media.address)) {
ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0);
media_address = &temp_media_address;
+ } else {
+ struct ast_sip_transport *transport =
+ ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport",
+ session->endpoint->transport);
+
+ if (transport && transport->state) {
+ char hoststr[PJ_INET6_ADDRSTRLEN];
+
+ pj_sockaddr_print(&transport->state->host, hoststr, sizeof(hoststr), 0);
+ ast_debug(1, "Transport: %s bound to host: %s, using this for media.\n",
+ session->endpoint->transport, hoststr);
+ ast_sockaddr_parse(media_address, hoststr, 0);
+ }
+ ao2_cleanup(transport);
}
if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->media.rtp.engine, sched, media_address, NULL))) {
@@ -217,7 +231,7 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
}
if (!strcmp(session_media->stream_type, STR_AUDIO) &&
- (session->endpoint->media.tos_audio || session->endpoint->media.cos_video)) {
+ (session->endpoint->media.tos_audio || session->endpoint->media.cos_audio)) {
ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_audio,
session->endpoint->media.cos_audio, "SIP RTP Audio");
} else if (!strcmp(session_media->stream_type, STR_VIDEO) &&