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authorJenkins2 <jenkins2@gerrit.asterisk.org>2017-09-28 12:32:49 -0500
committerGerrit Code Review <gerrit2@gerrit.digium.api>2017-09-28 12:32:49 -0500
commit88372c784470873a51fa31bc39fc5217602f574b (patch)
tree920c335b2d86997e742fab7975837b21ddaeef2f
parent3480e285b1081266c65f69942e8649e84f44faf6 (diff)
parent3d658e3be4de12f2594ade2572c19554f26ee82e (diff)
Merge "res_pjsip_session: outgoing call did not offer all configured codecs" into 15
-rw-r--r--res/res_pjsip_session.c10
1 files changed, 10 insertions, 0 deletions
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 9636583d2..70c88a504 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -2486,6 +2486,16 @@ struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint
continue;
}
+ if (ast_stream_get_type(req_stream) == AST_MEDIA_TYPE_AUDIO) {
+ /*
+ * By appending codecs from the endpoint after compatible ones this
+ * guarantees that priority is given to those while also allowing
+ * translation to occur for non-compatible.
+ */
+ ast_format_cap_append_from_cap(joint_cap,
+ endpoint->media.codecs, AST_MEDIA_TYPE_AUDIO);
+ }
+
ast_stream_set_formats(clone_stream, joint_cap);
ao2_ref(joint_cap, -1);