mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-03 19:16:46 +00:00
Merge "res_pjsip_sdp_rtp.c: Fix cut-n-paste error" into 13
This commit is contained in:
@@ -231,7 +231,7 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
|
|||||||
}
|
}
|
||||||
|
|
||||||
if (!strcmp(session_media->stream_type, STR_AUDIO) &&
|
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,
|
ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_audio,
|
||||||
session->endpoint->media.cos_audio, "SIP RTP Audio");
|
session->endpoint->media.cos_audio, "SIP RTP Audio");
|
||||||
} else if (!strcmp(session_media->stream_type, STR_VIDEO) &&
|
} else if (!strcmp(session_media->stream_type, STR_VIDEO) &&
|
||||||
|
Reference in New Issue
Block a user