[jitsi-users] Video ?


#1

Hi all,

Perhaps I'm trying something that's not implemented...
Between two jitsi-clients I can make a jingle call, audio and video
On those same machines I can only make SIP-audio calls

On those same machine I can make SIP-video call when I use at both ends another client, like linphone.

With jitsi, asterisks complains:
"ignoring video stream offer, because port number is zero"

When looking at the config, I noticed that only for sip and ssip ports are defined (5060 and 5061),
While for linphone I specify: sip, audio and video (5060, 7078, 9078)

Am I looking at the wrong places or missing something else....

Hans

···

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This message may contain information that is not intended for you. If you are not the addressee or if this message was sent to you by mistake, you are requested to inform the sender and delete the message. The State accepts no liability for damage of any kind resulting from the risks inherent in the electronic transmission of messages.


#2

Hey Hans

Hi all,

Perhaps I'm trying something that's not implemented...
Between two jitsi-clients I can make a jingle call, audio and video
On those same machines I can only make SIP-audio calls

On those same machine I can make SIP-video call when I use at both ends another client, like linphone.

I believe Linphone comes with H.263 enabled by default. You can try
enabling H.264 on your Asterisk or adding H.263 on Jitsi, if you are OK
with the lower quality.

With jitsi, asterisks complains:
"ignoring video stream offer, because port number is zero"

Yup, Jitsi would set the port of a media stream to 0 when it supports
none of the formats available for that stream.

Hope this helps,
Emil

···

On 01.10.12, 16:36, J.Witvliet@mindef.nl wrote:

When looking at the config, I noticed that only for sip and ssip ports are defined (5060 and 5061),
While for linphone I specify: sip, audio and video (5060, 7078, 9078)

Am I looking at the wrong places or missing something else....

Hans

______________________________________________________________________
Dit bericht kan informatie bevatten die niet voor u is bestemd. Indien u niet de geadresseerde bent of dit bericht abusievelijk aan u is toegezonden, wordt u verzocht dat aan de afzender te melden en het bericht te verwijderen. De Staat aanvaardt geen aansprakelijkheid voor schade, van welke aard ook, die verband houdt met risico's verbonden aan het elektronisch verzenden van berichten.

This message may contain information that is not intended for you. If you are not the addressee or if this message was sent to you by mistake, you are requested to inform the sender and delete the message. The State accepts no liability for damage of any kind resulting from the risks inherent in the electronic transmission of messages.

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
Jitsi
emcho@jitsi.org PHONE: +33.1.77.62.43.30
https://jitsi.org FAX: +33.1.77.62.47.31


#3

Hi Emil,

To exclude another component from the equation:
In next attempt, I dial into the echo function of asterisk (so no other phone/config involved)
Within sip.conf, I allow h264, h263 and h263p.

When using (same account settings) linphone or bria, the (video) echo works, but not for jitsi

Also when doing a sip-call between two jitsi's: even then it is audio only.
Tried jitsi with 1.0-3967 and 1.1-4126

hans

···

-----Original Message-----

From: Emil Ivov [mailto:emcho@jitsi.org]

Sent: Monday, October 01, 2012 11:48 PM
To: users@jitsi.java.net
Cc: Witvliet, J, CDC/IV/DCOPS/I&S/HIN
Subject: Re: [jitsi-users] Video ?

Hey Hans

On 01.10.12, 16:36, J.Witvliet@mindef.nl wrote:

Hi all,

Perhaps I'm trying something that's not implemented...
Between two jitsi-clients I can make a jingle call, audio and video
On those same machines I can only make SIP-audio calls

On those same machine I can make SIP-video call when I use at both ends another client, like linphone.

I believe Linphone comes with H.263 enabled by default. You can try
enabling H.264 on your Asterisk or adding H.263 on Jitsi, if you are OK
with the lower quality.

With jitsi, asterisks complains:
"ignoring video stream offer, because port number is zero"

Yup, Jitsi would set the port of a media stream to 0 when it supports
none of the formats available for that stream.

Hope this helps,
Emil

When looking at the config, I noticed that only for sip and ssip ports are defined (5060 and 5061),
While for linphone I specify: sip, audio and video (5060, 7078, 9078)

Am I looking at the wrong places or missing something else....

Hans

______________________________________________________________________
Dit bericht kan informatie bevatten die niet voor u is bestemd. Indien u niet de geadresseerde bent of dit bericht abusievelijk aan u is toegezonden, wordt u verzocht dat aan de afzender te melden en het bericht te verwijderen. De Staat aanvaardt geen aansprakelijkheid voor schade, van welke aard ook, die verband houdt met risico's verbonden aan het elektronisch verzenden van berichten.

This message may contain information that is not intended for you. If you are not the addressee or if this message was sent to you by mistake, you are requested to inform the sender and delete the message. The State accepts no liability for damage of any kind resulting from the risks inherent in the electronic transmission of messages.

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
Jitsi
emcho@jitsi.org PHONE: +33.1.77.62.43.30
https://jitsi.org FAX: +33.1.77.62.47.31


#4

Well, exclude Asterisk then because it likely puts itself in the way of the media. For example, put two Jitsi instances in the same network (because we do not support ICE for SIP) and use a provider who is just a SIP registrar and does not act as a media relay (e.g. iptel.org) on both of them.

···

On 03.10.2012, at 15:57, <J.Witvliet@mindef.nl> wrote:

To exclude another component from the equation:
In next attempt, I dial into the echo function of asterisk (so no other phone/config involved)
Within sip.conf, I allow h264, h263 and h263p.


#5

And having a look at the logs might also help :slight_smile:

···

On 03.10.12, 15:37, Lyubomir Marinov wrote:

On 03.10.2012, at 15:57, <J.Witvliet@mindef.nl> wrote:

To exclude another component from the equation:
In next attempt, I dial into the echo function of asterisk (so no other phone/config involved)
Within sip.conf, I allow h264, h263 and h263p.

Well, exclude Asterisk then because it likely puts itself in the way of the media. For example, put two Jitsi instances in the same network (because we do not support ICE for SIP) and use a provider who is just a SIP registrar and does not act as a media relay (e.g. iptel.org) on both of them.

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
Jitsi
emcho@jitsi.org PHONE: +33.1.77.62.43.30
https://jitsi.org FAX: +33.1.77.62.47.31