mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-12 15:45:18 +00:00
res_pjsip_sdp_rtp.c: Fix cut-n-paste error
We were inadvertenly referencing the cos_video option to determine if we should set the tos_audio and cos_audio value on the RTP instance. Change-Id: Ia7964f486801d39dc6f5dae570baff079e1595b0
This commit is contained in:
@@ -217,7 +217,7 @@ 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) &&
|
||||
|
Reference in New Issue
Block a user