PJSIP trunk codec configuration (GUI) filters Video CODECs


(Grand Poobah) #1

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 there’s a work around for this???

Thanks in advance

GPB1


How to set CPS (call per second) limit?
#2

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


(Grand Poobah) #3

Thanks Bill,

This is something that i’d like to have working. I can’t believe there isn’t a work around for this - not cool :frowning:


#4

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 “append” 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.


(Dave Burgess) #5

This looks like a job for a Feature Request with Code…


(Grand Poobah) #6

Bill,

Thank you very much for the input. I’ll definitely give this a go and may ping you & Cynjut for additional info if need be ( that’s if you don’t mind).

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

GPB1


(Dave Burgess) #7

I don’t speak for the company or the brand, but soon isn’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 The chan-sip code is old and is no longer actively maintained.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 ‘stupid’ that ChanSIP used to do exclusively is now “mostly” supported in PJ-SIP. It’s almost a full replacement for ChanSip and as soon as no one can find any more edge cases, it will go away.


#8

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

I don’t think the issue here is with pjsip as much as that video seems to be more of an afterthought in FreePBX. I don’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 cynjut’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).


(Grand Poobah) #9

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, I’ll take the information you’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


(Dave Burgess) #10

One thing that may not be crystal clear:

FreePBX isn’t Asterisk. It’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. It’s open source software, so if there’s things we want added, we need to add them.

I’ve only ever donated one thing to the project, and it was adopted “sideways”. 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.

That’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 else’s portfolio (mine is still out there, but please don’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, it’s a lot more likely that your “single person” change will make it into the system.


(Grand Poobah) #11

I used to code in Mesozoic languages like COBOL, RPG, FORTRAN & PASCAL in the 80’s but gave that up in the 90’s. I’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.


(Grand Poobah) #12

I almost for got -> thanks again for your & Bill’s input.


(Dave Burgess) #13

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. They’re pretty close. It’s not like we’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.


(Grand Poobah) #14

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


(system) closed #15

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