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| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-12-26 04:34:07 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-12-26 04:34:07 +0000 |
| commit | 1aba4079bcb5c22aa17a0502152c73764e1b538a (patch) | |
| tree | 756df2c8bb71afc320d31e4d30afe941fb099ef6 /channels/chan_sip.c | |
| parent | 7f137ceb7c311430a54fe9ebc3fdf3b2bb85cf8a (diff) | |
Merged revisions 48964 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48965 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/chan_sip.c')
| -rw-r--r-- | channels/chan_sip.c | 2 |
1 files changed, 2 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index e21e9670a..b02d0a7e1 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -4800,10 +4800,12 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) /* Initialize the temporary RTP structures we use to evaluate the offer from the peer */ newaudiortp = alloca(ast_rtp_alloc_size()); memset(newaudiortp, 0, ast_rtp_alloc_size()); + ast_rtp_new_init(newaudiortp); ast_rtp_pt_clear(newaudiortp); newvideortp = alloca(ast_rtp_alloc_size()); memset(newvideortp, 0, ast_rtp_alloc_size()); + ast_rtp_new_init(newvideortp); ast_rtp_pt_clear(newvideortp); /* Update our last rtprx when we receive an SDP, too */ |