sysmocom
/
asterisk
11
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

Only deal with getting the supported payloads on audio if an audio RTP stream exists 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp 2006-07-24 03:42:27 +00:00
parent fb94a4933b
commit 6810884f60
1 changed files with 3 additions and 2 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

5
channels/chan_sip.c
View File

@ -4447,7 +4447,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
int old = 0;
/* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */
int peercapability, peernoncodeccapability;
int peercapability = 0, peernoncodeccapability = 0;
int vpeercapability = 0, vpeernoncodeccapability = 0;
struct sockaddr_in sin; /*!< media socket address */
struct sockaddr_in vsin; /*!< Video socket address */
@ -4825,7 +4825,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
}
/* Now gather all of the codecs that we are asked for: */
ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
if (p->rtp)
ast_rtp_get_current_formats(newaudiortp, &peercapability, &peernoncodeccapability);
if (p->vrtp)
ast_rtp_get_current_formats(newvideortp, &vpeercapability, &vpeernoncodeccapability);