Summary: | ASTERISK-27883: No voice while using sipml5 in IPv6 environment | ||
Reporter: | Mohit Dhiman (mohitdhiman) | Labels: | IPv6 pjsip sipml5 webrtc |
Date Opened: | 2018-05-30 05:50:55 | Date Closed: | 2020-01-14 11:13:58.000-0600 |
Priority: | Minor | Regression? | |
Status: | Closed/Complete | Components: | Channels/chan_sip/IPv6 |
Versions: | 13.16.0 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | Centos 7 IPv6 | Attachments: | |
Description: | while using sipml5 along with asterisk 13 in an IPv6 environment i am not able to hear the playback that is configured in extensions.conf.
i did all the required configuration for IPv6 as: ;sip.conf [general] bindport=5060 bindaddr=[::] rtptimeout=300 rtpholdtimeout=600 transport=udp disallow=all allow=alaw,ulaw [1060] type=friend username=1060 secret=1060 host=dynamic encryption=yes avpf=yes icesupport=yes context=sipjs directmedia=no transport=udp,ws,wss force_avp=yes dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlssetup=actpass rtcp_mux=yes disallow=all allow=alaw,ulaw ;extensions.conf exten => 1061,1,Playback(demo-congrats) ;http.conf [general] enabled=yes bindaddr=[::] bindport=8088 tlsenable=yes tlsbindaddr=[::]:8089 tlscertfile=/etc/asterisk/keys/asterisk.pem | ||
Comments: | By: Asterisk Team (asteriskteam) 2018-05-30 05:50:56.993-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Joshua C. Colp (jcolp) 2018-05-30 05:54:18.608-0500 You need to always be using the latest version of Asterisk when using WebRTC, as things can change. As well I don't believe chan_sip has support for IPv6 ICE/STUN which would be required for this. I know it does work when using PJSIP. By: Mohit Dhiman (mohitdhiman) 2018-05-30 06:09:28.523-0500 When i am using twilio instead of sipml5 then it is working fine By: Joshua C. Colp (jcolp) 2018-05-30 06:14:09.457-0500 That isn't really the same thing. When using Twilio you are doing WebRTC between Twilio and your browser. When using sipml5 you are doing WebRTC between Asterisk and your browser. Twilio takes care of things in their case. By: Joshua C. Colp (jcolp) 2018-06-05 04:41:31.829-0500 Please try with chan_pjsip as like I previously stated, I do not believe this is supported by chan_sip. By: Asterisk Team (asteriskteam) 2018-06-19 12:00:01.664-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |