| Summary: | ASTERISK-24269: Delay in connection of speech with pjsip and webrtc | ||
| Reporter: | Abhay Gupta (agupta) | Labels: | |
| Date Opened: | 2014-08-26 01:53:21 | Date Closed: | 2014-09-16 09:47:14 |
| Priority: | Minor | Regression? | |
| Status: | Closed/Complete | Components: | Resources/res_pjsip |
| Versions: | SVN | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | ubuntu 12.04 with asterisk SVN-branch-12-r421978 | Attachments: | |
| Description: | Whenever bridge is attempted a error for stun is shown on CLI
ERROR[25051]: pjsip:0 <?>: icess0x7f4f140 ..Error sending STUN request: Invalid argument which probably is causing a delay in speech when the remote party answers . Rest all other functionality is working fine . {noformat} -- Executing [101@default:1] NoOp("PJSIP/102-00000004", "") in new stack -- Executing [101@default:2] Dial("PJSIP/102-00000004", "PJSIP/101") in new stack -- Called PJSIP/101 -- PJSIP/101-00000005 is ringing -- PJSIP/101-00000005 answered PJSIP/102-00000004 -- Channel PJSIP/102-00000004 joined 'simple_bridge' basic-bridge <df577e1b-673d-4b71-815b-62420e73783b> -- Channel PJSIP/101-00000005 joined 'simple_bridge' basic-bridge <df577e1b-673d-4b71-815b-62420e73783b> [Aug 26 12:06:29] ERROR[25051]: pjsip:0 <?>: icess0x7f4f140 ..Error sending STUN request: Invalid argument [Aug 26 12:06:29] ERROR[25051]: pjsip:0 <?>: icess0x1fdfeb8 ..Error sending STUN request: Invalid argument > 0x7f4f14041290 -- Probation passed - setting RTP source address to 192.168.1.243:62186 > 0x7f4f1402bea0 -- Probation passed - setting RTP source address to 192.168.1.161:58468 -- Channel PJSIP/101-00000005 left 'simple_bridge' basic-bridge <df577e1b-673d-4b71-815b-62420e73783b> -- Channel PJSIP/102-00000004 left 'simple_bridge' basic-bridge <df577e1b-673d-4b71-815b-62420e73783b> {noformat} | ||
| Comments: | By: Rusty Newton (rnewton) 2014-08-29 18:11:16.689-0500 [Edit] Actually - please provide your pjsip.conf configuration, rtp.conf configuration, a packet capture showing the call and an Asterisk debug log. Instructions on providing the debug log [can be found at this link|https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information]. By: Rusty Newton (rnewton) 2014-09-16 09:47:44.229-0500 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines | ||