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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 05:50:55Date Closed:2020-01-14 11:13:58.000-0600
Priority:MinorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/IPv6
Versions:13.16.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Centos 7 IPv6Attachments:
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