Unable to get video working

KCITS

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#1
Hi, I hope this is not too much of a newbie question but I have been battling for days trying to get video working with either X-lite or bria. The free versions were crashing so I tried version 4 beta and eventually paid for a version of bria 3.

I keep getting the message "Waiting for remote Video"

I have searched for and followed instructions in many posts most relating to asterisk or trixbox, but this does not help.

I have added the below lines in sip_custom.conf, in fact in frustration I have tried adding it to several sip_ conf files, but still get the same message.

videosupport=yes
maxcallbitrate=384
allow=g729
allow=g723
allow=h261
allow=h263
allow=h263p
allow=h264

What am I missing, perhaps there is a need to "allow video" for each extension.

At this stage I am just testing on a LAN between workstations with no firewall or router.
 

danardf

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#2
Hi.

First, prefer to put on every extension the video codec instead of general parameters.
videosupport=yes
maxcallbitrate=384
allow=g729 <- only if you have this codec installed.
allow=g723 <- only if you have this codec installed.
allow=h263
allow=h263p
Second, try to get the good Xlite version: V3.0 Build 53117 and disable the automatic update (dial a key sequence like ***7469), leave blank the field.
After this version, when you try to send a video, the Xlite breakdown with some error.

XLite, not use the H264 into the free verison, and the H261 is not use.
So, use only H263 & H263p.

Select the code into Xlite.

I use the video between XLite and a gxv3140.

Regards
 

KCITS

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#3
Hi
Sorry if I misunderstand, but by suggesting that I input this into each extension, do you mean I should append the sip_additional.conf by entering the below lines under each users extension?

As I have gone into PBX, extensions, and clicked on a users extension but can not find anywhere to enter the values you supplied.

PK
 

danardf

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#4
No, into every extension on Freepbx, you have the field allow
Into this field, you must put for example:
ulaw&h263&h263p

So:
disallow : all
allow: ulaw&h263&h263p

don't modify sip_additionnal.conf manually.

Every information into sip_general_custom.conf are used when there's no SIP information. So, it's only the SIP default parameters. But it's not very reliable. It's better to put the good parameters into every extension device. Like that, you could have a control on your config.
 

danardf

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#5
Else, you could read at first, Elastix Without Tears to know how to configure Elastix.

Regards
 

KCITS

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#6
Hi Franck
Thanks for your suggestions, we did scour the LWOTears pdf but did not find anything to assist with this problem.

I removed the entry from sip_custom.conf and have entered in the values,

disallow all
allow ulaw&h263&h263p

but still the message "waiting for remote video" is displayed, i can only see the local camera image in the small window.

I searched the logs in /var/logs/asterisk to see if there are any clues but I did not find any reference to these codecs or failures, I only see that the call was successfully established.
 

KCITS

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#7
Hi
Restarted asterisk service and i was unable to start video at all until i re-entered :

videosupport=yes
maxcallbitrate=384
allow=g729
allow=g723
allow=h261
allow=h263
allow=h263p
allow=h264

into sip_custom.conf, not sure if I need all these values but, now although I can activate video again there is still no remote video displayed.

Any additional help would be greatly appreciated.
 

danardf

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#8
You did put these codecs into Freepbx on every extension using the video?
Verifiy the call with this cmd (CLI).
sip show channels
and look at the codecs used.

Code:
sip show channels
Peer             User/ANR         Call ID      Seq (Tx/Rx)  Format           Hold     Last Message
193.107.20.92    104              26bc0a7b799  00102/00000  0x8 (alaw)       No       Tx: ACK
193.107.20.66    122              YmU5NDlkNzJ  00101/00002  0x2 (gsm)        No       Rx: ACK
 

KCITS

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#9
Hi
I entered the values into Elastix web interface but confirmed they were also in the freepbx, extension, details also

I executed the command and returned
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
192.168.47.69 8000 30a8a03f74f 00102/00000 0x3c0004 (ulaw| No Tx: ACK
192.168.47.29 1000 YTMyMWNhYjE 00101/00003 0x4 (ulaw) No Rx: ACK
2 active SIP channels
 

danardf

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#10
If I talk about sip_general_custom.conf, I did never talk about sip_custom.conf!
[*]videosupport=yes
[*]maxcallbitrate=384

and use only these codecs for the video.
allow=h261
allow=h263
allow=h263p

These parameters need to put only into the general section and not anywhere.

Of course, if you modify a paramter into a config file, you must reload asterisk, So, 2 way!
CLI> restart now or
CLI> reload

Please, read the document Elastix Whithout Tears. ;)

An example with my config file: (sip_general_custom.conf).
Code:
videosupport=yes
disallow=all
allow=alaw
allow=g729
canreinvite=no
nat=yes
language=fr
defaultexpirey=1800
dtmfmode=auto
qualify=yes 
srvlookup=yes
notifyringing=yes
notifybusy=yes
notifyhold=yes
limitonpeers=yes
subscribecontext=blf
call-limit=5
jbenable=yes
jbforce=no
jbmaxsize = 200                ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000
tos_sip=cs3                    ; Sets TOS for SIP packets.
tos_audio=ef                   ; Sets TOS for RTP audio packets.
tos_video=af41                 ; Sets TOS for RTP video packets.
And an example config extension:
Code:
[122]
deny=0.0.0.0/0.0.0.0
disallow=all
secret=*********
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=1
pickupgroup=1
allow=gsm
allow=h263
dial=SIP/122
accountcode=Distants
mailbox=122@default
permit=0.0.0.0/0.0.0.0
callerid=device <122>
call-limit=50
You could see that I have 2 codecs:
One for the audio codec and the other for the video codec.
 

danardf

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#11
KCITS said:
Hi
I entered the values into Elastix web interface but confirmed they were also in the freepbx, extension, details also

I executed the command and returned
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
192.168.47.69 8000 30a8a03f74f 00102/00000 0x3c0004 (ulaw| No Tx: ACK
192.168.47.29 1000 YTMyMWNhYjE 00101/00003 0x4 (ulaw) No Rx: ACK
2 active SIP channels
Yes so, you've only one audio codec.
Put your video codec into you extension 8000 and 1000.
 

KCITS

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#12
Hi Franck

thanks for your continued support, i know this must be frustrating for you but i did enter the forum throught he newbie door :)

I thought that I had entered the video codec into the extension by adding the string "ulaw&h263&h263p" in the allow section of the extensions properties? I think I must be mising something

Let me re-cap

I enter the lines
videosupport=yes
maxcallbitrate=384
allow=h261
allow=h263
allow=h263p

Into sip_general_custom.conf, and only here. (btw this file was empty).

Then enter in the allow section for the extension ulaw&h263&h263p as a single line.
enter sip show channels in the CLI but only see ulaw

view sip_additional.conf
the two extensions look like this

[1000]
deny=0.0.0.0/0.0.0.0
disallow=all
type=friend
secret=********
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=1000@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/1000
context=from-internal
canreinvite=no
callgroup=
callerid=device <1000>
allow=ulaw
allow=h263
allow=h263p
accountcode=
call-limit=50

[8000]
deny=0.0.0.0/0.0.0.0
type=friend
secret=****
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=8000@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/8000
context=from-internal
canreinvite=no
callgroup=
callerid=device <8000>
allow=ulaw
allow=h263
allow=h263p
accountcode=
call-limit=50

To me this seems to be all ok, with the exception that my sip_general_custom.conf is very bland in comparison to yours, there is no tos settings for sip, audio or video is this relevant? but there is still no remote video. I am stumped
 

danardf

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#13
I go away for some hours, I come back later.

Remember that you must reload Asterisk after every modification!

You must select your video codec into you Xlite config too. Right clic (Options / Advenced / Video Codecs).

Letme know.

Regards
 

KCITS

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#14
Hi
I have copied below the output sip show channel for the two extensions, i can not specifically see in there that videosupport is allowed, but do see the codecs in the 8000 extension only. I will work on this and RTFM very closely :) . I will copy any progress to this forum.

CLI sip show channel NDQ1YWRjNWR
* SIP Call
Curr. trans. direction: Incoming
Call-ID: NDQ1YWRjNWRkZWFjZTgxMzAzMWMwZDQ5ZTkwNzAzZGQ.
Owner channel ID: SIP/1000-00000007
Our Codec Capability: 1572868
Non-Codec Capability (DTMF): 1
Their Codec Capability: 3670284
Joint Codec Capability: 1572868
Format: 0x4 (ulaw)
MaxCallBR: 384 kbps
Theoretical Address: 192.168.47.29:61230
Received Address: 192.168.47.29:61230
SIP Transfer mode: open
NAT Support: Always
Audio IP: 192.168.47.5 (local)
Our Tag: as6e966cd0
Their Tag: 8e5d0d23
SIP User agent: Bria 3.0 release 3.0.1.1 stamp 56993
Username: 1000
Peername: 1000
Original uri: sip:1000@192.168.47.29:61230
Caller-ID: 1000
Need Destroy: 0
Last Message: Rx: ACK
Promiscuous Redir: No
Route: sip:1000@192.168.47.29:61230
DTMF Mode: rfc2833
SIP Options: eventlist

CLI sip show channel 101272cc691
SIP Call
Curr. trans. direction: Outgoing
Call-ID: 101272cc691fe9e87d1ccb9572c0e7a3@192.168.47.5
Owner channel ID: SIP/8000-00000008
Our Codec Capability: 1835020
Non-Codec Capability (DTMF): 1
Their Codec Capability: 1572876
Joint Codec Capability: 1572876
Format: 0x1c0004 (ulaw|h261|h263|h263p)
MaxCallBR: 384 kbps
Theoretical Address: 192.168.47.69:20346
Received Address: 192.168.47.69:20346
SIP Transfer mode: open
NAT Support: Always
Audio IP: 192.168.47.5 (local)
Our Tag: as55a41266
Their Tag: b05da798
SIP User agent: X-Lite Beta release 4.0 Beta 2 stamp 56233
Username: 8000
Peername: 8000
Original uri: sip:8000@192.168.47.69:20346
Need Destroy: 0
Last Message: Tx: ACK
Promiscuous Redir: No
Route: sip:8000@192.168.47.69:20346;rinstance=184ad37196532a8a
DTMF Mode: rfc2833
SIP Options: (none)
 

KCITS

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#15
one thing I have noted and I am not sure if this is relevent

Whichever extension receives the call has the below codecs listed when executing "sip show channel *******"

Format: 0x1c0004 (ulaw|h261|h263|h263p)

The extension which intitiates the call displays

Format: 0x4 (ulaw)
 

danardf

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#16
Yet here, you have a H261 that's not supported by XLite.
 

KCITS

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#17
I have removed the H261 codec, only leaving the below
Format: 0x180004 (ulaw|h263|h263p)

unfortunately I still still get the message "waiting for remote video" on all extensions.

I am not sure where to go to diagnose further, I have tested it on a fresh VM build with the same result.

Is this perhaps broken in Elastix 1.6.2-2
 

KCITS

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#18
The problem appears to be with counterpaths X-Lite and Bria softphones. In frustration we installed BOL Sip Phone on one PC.

When the call is initiated from this application the video works correctly at both ends, although when we return a call to the “BOL Sip Phone” from either Xlite or Bria, the video is still unable to be displayed at either end.

Has anyone else seen this problem?

Anyway, I have emailed their support and I will post their reply, and fix (if received) back to this forum.
 

danardf

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#19
Hi. I said you about these versions there was some video problems . You should try to take a specific version V3.0 Build 53117.

You can try to find it on google with only => X-Lite_Win32_1103d_53117 and take the result. Lots of link are there!

With this version I've not any problem.

If there's a video problem with your verison, yes you must contact the counterpath support.
 

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