| Summary: | ASTERISK-22415: asterisk-11.5.0 linohone 3.6.1 ice not work | ||
| Reporter: | wangpeng (voipwangpeng) | Labels: | |
| Date Opened: | 2013-08-27 20:31:18 | Date Closed: | 2013-08-28 08:45:13 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Resources/res_rtp_asterisk |
| Versions: | 11.5.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | my asterisk have a domain name with dynamic ipaddress.
i want to use asterisk and linphone make p2p call, i open icesupport in sip.conf and rtp.conf, can call through but no voice. here https://wiki.asterisk.org/wiki/display/~jcolp/ICE,+STUN,+and+TURN+Support i see ICE support is only used for communication between a remote endpoint and Asterisk. so i close icesupport in sip.conf and rtp.conf, but invite from caller have ice paramter,but asterisk modify invite message, delete ice paramter, then pass to callee, so phenomenon is also can call through but no voice. how to make asterisk not modify sdp of invite, or have other method to use asterisk and linphone make p2p call. | ||
| Comments: | By: Rusty Newton (rnewton) 2013-08-28 08:45:01.190-0500 Thanks for your comments. This does not appear to be a bug report and we are closing it. We appreciate the difficulties you are facing, but it would make more sense to raise your question to the community before filing an issue: http://www.asterisk.org/community/discuss Before filing an issue in the future, please read through the guidelines: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines After seeking help in the community. If you feel you have a bug, remember to post the required debug information to demonstrate such an issue. See the above guidelines. | ||