diff options
author | Kinsey Moore <kmoore@digium.com> | 2014-06-06 19:13:08 +0000 |
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committer | Kinsey Moore <kmoore@digium.com> | 2014-06-06 19:13:08 +0000 |
commit | 5510e3c699a6f0e6318bbfe4f27b241f42e4c2dc (patch) | |
tree | 34e4bf0c02a7f8103d31d376d3cdf5b49d418775 /res | |
parent | 077c4187d9789eaf585568f8178f5e3a470ab781 (diff) |
PJSIP: Remove premature write of raw formats
Currently, there are situations that can occur when using chan_pjsip
and certain dialplan applications (notably ChanSpy()) that can cause
the channel to get no audio with scrolling warnings about format
mismatches. This is caused by a failure to update translation paths on
a mid-call native format update since the raw formats have already
been updated by res_pjsip_sdp_rtp.c in set_caps(). Removing the
premature raw format updates allows the translation paths to be setup
correctly and the raw read and write formats with them.
AFS-63 #close
........
Merged revisions 415342 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res')
-rw-r--r-- | res/res_pjsip_sdp_rtp.c | 2 |
1 files changed, 0 insertions, 2 deletions
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index 43f0832ac..59fa647d9 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -251,8 +251,6 @@ static int set_caps(struct ast_sip_session *session, struct ast_sip_session_medi /* Apply the new formats to the channel, potentially changing read/write formats while doing so */ ast_format_cap_copy(ast_channel_nativeformats(session->channel), caps); - ast_format_copy(ast_channel_rawwriteformat(session->channel), &fmt); - ast_format_copy(ast_channel_rawreadformat(session->channel), &fmt); ast_set_read_format(session->channel, ast_channel_readformat(session->channel)); ast_set_write_format(session->channel, ast_channel_writeformat(session->channel)); } |