| Summary: | ASTERISK-29532: Issue with maxptime and VolTE (Voice Over LTE) | ||
| Reporter: | Renato Ribas (ribascr) | Labels: | |
| Date Opened: | 2021-07-27 16:04:46 | Date Closed: | |
| Priority: | Minor | Regression? | |
| Status: | Open/New | Components: | Codecs/General |
| Versions: | 16.19.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Sangoma Linux release 7.8.2003 (Core) | Attachments: | |
| Description: | 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 | ||
| Comments: | By: Asterisk Team (asteriskteam) 2021-07-27 16:04:51.354-0500 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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. 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]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Joshua C. Colp (jcolp) 2021-07-27 16:09:29.291-0500 Can you provide the language or part of the RFC which actually states this? By: Renato Ribas (ribascr) 2021-07-27 16:18:55.435-0500 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 By: Renato Ribas (ribascr) 2021-07-27 16:28:40.145-0500 Updated the maxptime for better clarification. The sentence that previously read: "The time SHOULD be a multiple of the frame size." now says "The time SHOULD be an integer multiple of the frame size." This should have no impact on interoperability. maxptime: The maximum amount of media which can be encapsulated in a payload packet, expressed as time in milliseconds. The time is calculated as the sum of the time that the media present in the packet represents. The time SHOULD be an integer multiple of the frame size. If this parameter is not present, the sender MAY encapsulate any number of speech frames into one RTP packet. On the other hand, the transcoding from G.711 to UEMCLIP is not entirely straightforward. Since there are no means to generate enhancement sub-layers, a G.711 bitstream can only be converted to UEMCLIP Mode 0 bitstream. If the original G.711 bitstream is encoded in A-law, it should first be converted to u-law to become the core layer. Because a UEMCLIP frame size is 20 ms, a u-law-encoded G.711 bitstream MUST be a 160-sample chunk to become a core layer. I understand by definition that the maxptime value needs to be a multiple of 20. https://datatracker.ietf.org/doc/html/rfc4566 https://datatracker.ietf.org/doc/html/rfc5686 | ||