PJSIP trunk codec configuration (GUI) filters Video CODECs

PJSIP trunk configurations are filtering video CODECs ( H264 , VP8 etc) outbound.

I see the Video Codecs being forwarded by my soft client to the server and I have H264 & VP8 enabled under the Asterisk SIP settings configuration, as well as in the extension allowed field; however,
the GUI settings for the PJSIP trunk group ( CODECS ) are filtering the video CODECS from the INVITE request.

The INVITE forwarded from the server will conform to any CODEC settings defined for the PJSIP trunk group; but there appears to be no settings defined here to allow any video CODECs.

Does anyone know if thereā€™s a work around for this???

Thanks in advance

GPB1

An untouched feature request https://issues.freepbx.org/browse/FREEPBX-15853

Thanks Bill,

This is something that iā€™d like to have working. I canā€™t believe there isnā€™t a work around for this - not cool :frowning:

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Yes, there is a simple workaround. Which makes me wonder why this is not available in the web interface yet.

Edit /etc/asterisk/pjsip.endpoint_custom_post.conf

Add an ā€œappendā€ block (+) containing your options. e.g.:

[Zoom](+)
allow=h264

The name within the [ ] matches the name you used in the trunk config:

Save the .conf file and then use fwconsole r in the terminal to reload.

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This looks like a job for a Feature Request with Codeā€¦

Bill,

Thank you very much for the input. Iā€™ll definitely give this a go and may ping you & Cynjut for additional info if need be ( thatā€™s if you donā€™t mind).

Offhand, does anyone know when / the time frame when ChanSIP configs will no longer be supported?

GPB1

I donā€™t speak for the company or the brand, but soon isnā€™t too soon. The chan-sip code is old and is no longer actively maintained. No new features have been added in years and everything ā€˜stupidā€™ that ChanSIP used to do exclusively is now ā€œmostlyā€ supported in PJ-SIP. Itā€™s almost a full replacement for ChanSip and as soon as no one can find any more edge cases, it will go away.

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G-Poob: (may I call you that?)

I donā€™t think the issue here is with pjsip as much as that video seems to be more of an afterthought in FreePBX. I donā€™t blame anyone for this as FreePBX was not designed as a video platform; it was designed as a PBX.

There are other tickets in place for video options enhancements.

I agree with cynjutā€™s assessment. I would use pjsip and just disable the chan_sip driver. If you happen to find a case that only chan_sip can solve, you can enable it (and file a ticket with FreePBX or Asterisk to let them know).

Thanks Bill & Dave,

Again, I thoroughly appreciate your time and assistance. My initial testing was performed with chanSIP configurations in place ( endpoints / trunks etc) and there were no issues with the transmission and reception of voice or video. So, If there was a definitive timeline ( lets say 3-5 years) for the EOL support of ChanSIP configurations, I could possibly continue with my testing as is/was.

Regardless, Iā€™ll take the information youā€™ve provided to heart, and will continue with the conversion from a ChanSIP to PJSIP PBX environment - thanks again for the assist.

btw - G-Poob/GPB1 is just fine (What can I say, I loved watching the Flintstones when I was kid).

GPB1

One thing that may not be crystal clear:

FreePBX isnā€™t Asterisk. Itā€™s more like a management framework for Asterisk. Asterisk will get full support for video codecs long before FreePBX will, even though Digium and Sangoma are now the same commercial entity. Itā€™s open source software, so if thereā€™s things we want added, we need to add them.

Iā€™ve only ever donated one thing to the project, and it was adopted ā€œsidewaysā€. I wrote an SCCP Manager (back in the 1.6 days) and added some Report changes for Chan-SCCP support (lines, devices, etc.) so that you could look at Asterisk Info at a glance and get the scoop. Rather than adopt my singular changes, they decided to modify Asterisk Info so that more modules (Chan-GSM is a thing, after all) could report information through the AsteriskInfo module.

Thatā€™s how a lot of this stuff gets into the system. You need it, you make it happen, a couple other people like it and help it move forward, and eventually it gets support in FreePBX. The SCCP Manager program is now in someone elseā€™s portfolio (mine is still out there, but please donā€™t look) and it fully supports the current version of FreePBX. They really took it and ran with it in ways I could never afford to.

In other words, if you can figure out how to make something happen and provide code to make it happen, itā€™s a lot more likely that your ā€œsingle personā€ change will make it into the system.

I used to code in Mesozoic languages like COBOL, RPG, FORTRAN & PASCAL in the 80ā€™s but gave that up in the 90ā€™s. Iā€™d be completely lost if I tried my hand @ any of the newer languages (C# / C++ / JAVA) etc, but will need to wait for one of you geniuses to make that happen.

I almost for got -> thanks again for your & Billā€™s input.

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I love it when the kids drop in for a visit.

If you can do any work in Pascal, you should be able to grind your way through PHP. Theyā€™re pretty close. Itā€™s not like weā€™re building the phone systems with format statements in column 8 or dropping 77-levels into the logic. :slight_smile:

I have confidence - you can do it.

Thanks for the vote of confidence. If I can find the time outside of work & family, Iā€™ll give it a shot.

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