927473 - Add "From Legacy" services for SIP video to VidyoGateway

Status: RESOLVED FIXED

(Infrastructure & Operations Graveyard :: AVOps: Vidyo, task)

Description

[Dave Miller [:justdave]]

We have no video-capable services enabled in VidyoGateway that are compatible with the SIP trunk from Asterisk.  We'd like to be able to enable users to call in with a video-capable SIP client.

Right now I can't seem to make it work.  I'm pretty sure the reason is because we only have H.323 services defined for inbound, and not SIP.

The "06" "07" and "08" services for SIP are currently defined as "To Legacy" only.  Those should be cloned with identical settings except for the direction being "From Legacy".

I'm guessing (but not positive) that it'll want to restart to apply the settings changes, so it probably needs CAB approval and done after hours.

Comment 1

[Gervase Markham [:gerv]]

justdave: AIUI, fixing this bug would enable people to interact with the Vidyo system using free software clients. Is that correct? 

If so, do you or Guillermo have a progress report?

Gerv

Comment 2

[Dave Miller [:justdave]]

(In reply to Gervase Markham [:gerv] from comment #1)
> justdave: AIUI, fixing this bug would enable people to interact with the
> Vidyo system using free software clients. Is that correct? 

That is correct.

> If so, do you or Guillermo have a progress report?

Guess it's waiting on Guillermo at this point.

Comment 3

[Gervase Markham [:gerv]]

guillermo: we've had a new version of Vidyo since this bug was filed. What's the story with it interoperating with SIP audio and video clients?

Gerv

Comment 4

[Guillermo Huerta [Disabled MoCo account]]

Gateway versions haven't changed, but I've made the change.  Please give this a try when you get a chance.

Comment 5

[Gervase Markham [:gerv]]

justdave: how can I test this? This bug doesn't explain how one might call in to a Vidyo call with a SIP client. I tried e.g. sip:[e-mail], but no response.

Gerv

Comment 6

[Dave Miller [:justdave]]

Dialing extension 92 from a video-capable softphone logged into Mozilla's phone system should work.

Comment 7

[Dave Miller [:justdave]]

Or sip:[e-mail]  where xxx is the Vidyo conference room number (add the 9 in front)

Comment 8

[Gervase Markham [:gerv]]

justdave: I've tried the following SIP addresses in my desktop SIP client (Linphone), using a Sipgate account, and none of them work:

LON 341 Commons:
sip:[e-mail]
sip:[e-mail]
Gerv's Room:
sip:[e-mail]
sip:[e-mail]

Should these work?

My goal here, in case it's not obvious, is to allow people to use standards-compliant and open source software to dial in to Vidyo calls.

Can you get it working and let me know what config you used?

Gerv

Comment 9

[Dave Miller [:justdave]]

so it depends on the client ...  we have an SRV record in DNS pointing mozilla.com at pbx.mtv2.mozilla.com, however I've notice that YateClient, for example, doesn't look up the SRV record.  If you actually put pbx.mtv2.mozilla.com after the @ then it works (in YateClient).  Also essential to the process in this case is that you are not using an account.  It needs to be attempting direct SIP without the aid of an account provider.

Comment 10

[Gervase Markham [:gerv]]

Having switched to direct-dial SIP, I now get the following results:

* Anything with "conf" prepended gives "User not found" as a textual message in the phone's UI. Are you sure that part is needed?

* sip:[e-mail] gives a voice saying "I'm sorry, that is not a valid extension"
* sip:[e-mail] rings and rings. Is it ringing the actual phone in the conference room?
* sip:[e-mail] rings once then hangs up on me
* sip:[e-mail] rings once then hangs up on me

Please can you give exact steps (and SIP URLs) you used to reproduce success?

Gerv

Comment 11

[Gervase Markham [:gerv]]

And by "success", of course I mean "success connecting to a Vidyo conference room".

Gerv

Comment 12

[Dave Miller [:justdave]]

Yes, the "conf" part is needed, and that's what I used to successfully connect:
sip:[e-mail] (for the internal Asterisk conference room 237)
sip:[e-mail] (for the Vidyo conference room 237)

And without that, it does actually ring that extension if you put a valid extension there, assuming that extension is available to be dialed at the main menu if you call one of the outside phone numbers and use the IVR system.

We also have fail2ban watching the logs and it bans users for 6 hours if they get invalid extension 3 times (because it's commonly attacked), so you probably got banned.

Note that Vidyo gives you no audio indication that you've actually connected.  If there's nobody else already in there room, it'll just stop ringing and you'll hear silence.

Comment 13

[Dave Miller [:justdave]]

Using YateClient on the Mac:

Set Protocol to "sip", set Account to "none"
In the dial box, put in "[e-mail]"
Click "Call".

Comment 14

[Gervase Markham [:gerv]]

OK, I got audio working :-) Video still doesn't work. The call seems to pick H.263, which results in a light blue solid window in my Vidyo client. I can see a video preview locally, albeit that it only takes up a small bit of the preview window. 

Can anyone give some guidance as to which codecs this system should support?

The thing which was blocking it from working, it turns out, is that I had my Mozilla SIP account configured. Even if I tell Linphone not to register that account, the presence of this account in the config seems to break the ability to make a direct connection. (If I select that account as the account to use, it still doesn't work.) I assume it associates the two because the address I am dialling is at the same domain as the account. This may well be a Linphone bug.

I guess this problem will not be faced by most people trying to dial in this way, as they won't have accounts on our SIP server, so it's not going to be an issue.

I've updated
https://wiki.mozilla.org/Teleconferencing#Internet_calling_.28VoIP.29
- let me know if it's wrong.
 
Gerv

Comment 15

[Dave Miller [:justdave]]

I believe it's set up to support both H.263 and H.264, Guillermo will have to let me know if that's not correct.

If you're logged into your Mozilla softphone account, then you should be able to get video just dialing x92 instead of using the sip: dialstring.

Comment 16

[Guillermo Huerta [Disabled MoCo account]]

It is set to support both.

Comment 17

[Gervase Markham [:gerv]]

Can this bug be made public so I can point people at it and others can do some testing?

Gerv

Comment 18

[Dave Miller [:justdave]]

Yep, I see nothing confidential in here.

Comment 19

[Daniele "Mte90" Scasciafratte]

Maybe a wiki page with all the step for test and use this solution will be very useful.

Comment 20

[Gervase Markham [:gerv]]

Daniele: no-one is certain how to make it work reliably; that's what:
http://blog.gerv.net/2014/08/accessing-vidyo-meetings-using-free-software-help-needed/
is all about. If you know how to do it, please do write the wiki page, and give us a link here!

Gerv

Comment 21

[Dan Callahan [:callahad]]

I haven't gotten as far as Gerv, yet. Trying to simulate what an external contributor might use.

Context:
- Client: Jitsi (F/OSS, running on OS X)
- SIP Account: [e-mail] (Free, from the Ekiga softphone team)
- Mozilla VPN not connected
- I have a Mozilla softphone (extension 817) and a personal Vidyo room (extension 9814)

The SRV record seems fine; I get identical results when dialing both @mozilla.com and @pbx.mtv2.mozilla.com

Calling [e-mail] successfully rings my Mozilla-provided softphone.

Calling [e-mail] appears to connect and remain connected, but I don't see/hear the room, the room doesn't see/hear me. I don't appear in the list of participants in the Vidyo room management dashboard.

Calling [e-mail] dials, responds with recording: "That is not a valid extension. Please hang up and try again."

Comment 22

[Dan Callahan [:callahad]]

Ooh! Dialing [e-mail] did successfully connect me to the Vidyo room, once.

I was able to see other participants, and they could hear me. No video from my camera, but I suspect that's a client problem. Trying again with another SIP client...

Comment 23

[Dan Callahan [:callahad]]

Linphone works! Reliably!

The only setting I had to change was Options -> Preferences -> Codecs -> Video codecs -> H263 -> Enable

(Not H263-1998; normal H263)

Once that was done, I could dial into [e-mail] and everything (bidirectional A/V) worked.

I didn't even have to sign in with my Ekiga account -- the default dan@192.168.1.254 (local IP) account the Linphone tries to use worked just fine.

Comment 24

[Gervase Markham [:gerv]]

Dan: great! That's some of what justdave said in comments 12 and 13.

However, with the same settings, I can't connect to  [e-mail] - it just hangs at "Contacting". However, [e-mail] (my extension) works - sort of. (I get a very small video window, at any rate.) Do some of these rooms require an actual Vidyo user to be in them to exist, do you think?

Gerv

Comment 25

[Dan Callahan [:callahad]]

Attachment: linphone-3.7.0-works.jpg

Attaching a screenshot proving that this is working for me, with relevant settings visible.

I'll write up a WikiMo page next week, after I've experimented with other clients and operating systems.

Comment 26

[Gervase Markham [:gerv]]

callahad: did your write-up ever make it to a wiki page? It would be great if we could update https://wiki.mozilla.org/Teleconferencing with clear and working instructions on how to connect into Vidyo via SIP. That's been my goal...

Gerv

Comment 27

[Dave Miller [:justdave]]

(In reply to Dan Callahan [:callahad] from comment #22)
> Ooh! Dialing [e-mail] did successfully connect me to
> the Vidyo room, once.

For anyone who can't get the @mozilla.com SRV record to work, the direct hostname is now anonsip1.scl3.mozilla.com -- the mtv2 server no longer supports it.

Comment 28

[Dan Callahan [:callahad]]

(In reply to Gervase Markham [:gerv] from comment #26)
> callahad: did your write-up ever make it to a wiki page?

Unfortunately, I was only ever able to get it to work with LinPhone on OS X -- no other client I attempted to use on Mac OS X or Linux was able to successfully do video.

For audio, it was as simple as dialing sip:conf9<vidyo-ext>@mozilla.com and things seemed to Just Work, and those instructions are already on the Wiki.

Happy to take another run at testing stuff if it would be valuable.

Comment 29

[Dave Miller [:justdave]]

I had it working on X-Lite on OS X back then when we were testing it.

Comment 30

[Dave Miller [:justdave]]

(In reply to Dave Miller [:justdave] ([e-mail]) from comment #29)
> I had it working on X-Lite on OS X back then when we were testing it.

Oh, X-Lite doesn't do point-to-point SIP.  It works dialing x92 while logged into Mozilla's phone system though.

Comment 31

[Gervase Markham [:gerv]]

justdave: anonsip1.scl3.mozilla.com doesn't respond to ping, and Linphone debug says:

message: eXosip_dnsutils_naptr_lookup: About to ask for 'anonsip1.scl3.mozilla.com NAPTR'
error: eXosip_dnsutils_naptr_lookup: res_query failed ('anonsip1.scl3.mozilla.com NAPTR')

I get the same error when trying @mozilla.com too.

message: eXosip_dnsutils_naptr_lookup: About to ask for 'mozilla.com NAPTR'
error: eXosip_dnsutils_naptr_lookup: res_query failed ('mozilla.com NAPTR')

Are you sure this is right? dig suggests anonsip1.scl3.mozilla.com doesn't have a NAPTR, and neither does mozilla.com...

Gerv

Comment 32

[Dave Miller [:justdave]]

I had never heard of NAPTR before, wasn't aware anything needed it.  Everything I've touched before directly used the SRV record.  Yes, that's the correct domain name.

So obviously we need to set up that NAPTR record then.

The content of this record does *not* look very straightforward.  It's http://www.ietf.org/rfc/rfc2915.txt for future reference.

Comment 33

[Dave Miller [:justdave]]

bug 1128415 has been filed to get the NAPTR set up.

Comment 34

[Dave Miller [:justdave]]

Just to keep everyone up to date, bug 1128415 has been all but wontfixed as the system we use to manage our DNS does not currently support NAPTR records and trying to make it support them would be too large a task for a one-off like this.

Comment 35

[Gervase Markham [:gerv]]

(In reply to Dave Miller [:justdave] ([e-mail]) from comment #27)
> For anyone who can't get the @mozilla.com SRV record to work, the direct
> hostname is now anonsip1.scl3.mozilla.com -- the mtv2 server no longer
> supports it.

This is a typo! The correct hostname is anonsip.scl3.mozilla.com. Perhaps that's why people were having problems!

Gerv