Summary: | ASTERISK-24166: SDP Asterisk. | ||
Reporter: | abdasty (abdasty) | Labels: | |
Date Opened: | 2014-08-06 10:36:30 | Date Closed: | 2014-08-06 11:29:48 |
Priority: | Major | Regression? | |
Status: | Closed/Complete | Components: | Resources/res_rtp_asterisk |
Versions: | 11.11.0 | Frequency of Occurrence | |
Related Issues: | |||
Environment: | Debian GNU/Linux 7 \n \l Asterisk 11.11.0 built by root @ asterisk on a x86_64 running Linux on 2014-08-06 15:08:44 UTC System Statistics ----------------- System Uptime: 25 hours Total RAM: 8186580 KiB Free RAM: 3641452 KiB Buffer RAM: 227732 KiB Total Swap Space: 999420 KiB Free Swap Space: 999420 KiB Number of Processes: 251 | Attachments: | ( 0) full.ok |
Description: | Hello,
here an issue about SDP Asterisk. Asterisk debug 9 Asterisk verbose 9 Call flow is : SIP Softphone => Asterisk => PSTN Weh I make a call, I had the following errors : {noformat} [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing session-level SDP s=pjmedia... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing session-level SDP b=AS:84... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:1... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing media-level (audio) SDP b=TIAS:64000... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:23437 IN IP4 46.255.52.118... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing session-level SDP s=pjmedia... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing session-level SDP b=AS:84... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:1... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing media-level (audio) SDP b=TIAS:64000... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:63200 IN IP4 46.255.52.118... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=candidate:Pa280135 1 UDP 2130706431 46.255.52.118 58124 typ prflx raddr 10.40.1.53 rport 50113... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=candidate:Pa280135 2 UDP 2130706430 46.255.52.118 63200 typ prflx raddr 10.40.1.53 rport 50116... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=remote-candidates:1 62.210.74.52 17240 2 62.210.74.52 17241... UNSUPPORTED OR FAILED. [2014-08-06 17:30:35] DEBUG[25070][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED. {noformat} Please find in attachment debug files Regards. | ||
Comments: | By: abdasty (abdasty) 2014-08-06 10:37:55.372-0500 Debugs when I reproduce the problem By: Matt Jordan (mjordan) 2014-08-06 11:29:43.719-0500 Those aren't errors. They're merely the DEBUG output as Asterisk parses an SDP. In fact, the INVITE request was processed just fine and a 200 OK with a valid SDP was sent to the UA that initiated the request. In the future, if you have questions about what Asterisk is telling you, please use the asterisk-users mailing list or the #asterisk IRC channel. |