| Summary: | ASTERISK-23083: PJSip not honouring rtp port definitions | ||
| Reporter: | xrobau (xrobau) | Labels: | |
| Date Opened: | 2013-12-31 21:18:50.000-0600 | Date Closed: | 2013-12-31 21:25:42.000-0600 |
| Priority: | Major | Regression? | Yes |
| Status: | Closed/Complete | Components: | Channels/chan_pjsip |
| Versions: | 12.0.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | After noticing a no-audio issue when calling INTO a nat-ted PJSip machine, I discovered that it was responding with a RTP port that was outside of the range it should have been, in response to the INVITE.
rtp.conf consists of: [general] icesupport=yes rtpstart=25000 rtpend=29000 | ||
| Comments: | By: xrobau (xrobau) 2013-12-31 21:25:42.956-0600 It appears to be a configuration issue at my end, ignore. Will reopen/recreate if I duplicate it | ||