[asterisk-bugs] [JIRA] (ASTERISK-29532) Issue with maxptime and VolTE (Voice Over LTE)

[Renato Ribas (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29532?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=255707#comment-255707 ] 

Renato Ribas commented on ASTERISK-29532:
-----------------------------------------

I will look for the excerpt in the RFC, as the adjustment information was requested by the telecom operator.

As a test I performed the procedure of changing the source code and replacing the binary and this really solves the problem

cd /usr/src

wget http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-16.17.0.tar.gz

tar xvf asterisk-16.17.0.tar.gz

asterisk-16.17.0 cd

cd main

vim or nano codec_biltin.c - (Change maximum_ms = 120 to ulaw, alaw and maximum_ms = 220 to g729)

./configure --libdir=/usr/lib64 --with-jansson-bundled

make menuselect (Select Macro and Sounds EN and Save)

make

fw console stop

mv /usr/sbin/asterisk /usr/sbin/asterisk_rpm

cp /usr/src/asterisk-16.17.0/main/asterisk /usr/sbin/

fw console start

Thanks

> Issue with maxptime and VolTE (Voice Over LTE)
> ----------------------------------------------
>
>                 Key: ASTERISK-29532
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29532
>             Project: Asterisk
>          Issue Type: New Feature
>      Security Level: None
>          Components: Codecs/General
>    Affects Versions: 16.19.0
>         Environment: Sangoma Linux release 7.8.2003 (Core)
>            Reporter: Renato Ribas
>            Assignee: Renato Ribas
>
> Hello how are you?
> Some telecom providers have problems with outgoing calls over the SIP trunk if destines using VoLTE (Voice Over LTE).
> According to diagnosis, the value of maxptime=150, however, according to RFC 4566, it must be a multiple of 20.
> The parameter is fixed in the asterisk code in codec_builtin.c, when performing the manual compilation I lose some functionality of commercial modules like Sysadmin in FreePBX.
> Would it be possible in future versions for the parameter to be pre-compiled according to RFC?
> Thanks



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