[sip-comm-dev] jingle videocalls testing successful


#1

Dear all,
want to share my experience with SC-videocalls using jingle (Gmail) between two windows-labtops (nb 3081) calling Berlin-Vienna:

nice video (no delay in time)
audio ok (a bit jittering)
zRTP is working!

Congratulations! THIS IS REALLY GREAT! :slight_smile:

kind regards, MS

路路路

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#2

Hi All,

Yes i agree it is really great in terms of performances, but P2P network condition is not always possible, and so there is no audio or one way audio in many conditions, is there any thing that is possible to do to discover P2P network condition or relay RTP in some cases ? I know this is one of the limitation of Jingle to be a mass production solution, but may be there is some solution for it recently ?

It will be great to have more details regarding mass deploying of the solution

Regards
Fabio Galdi

路路路

Il giorno 16/nov/2010, alle ore 18.25, Mr Smith ha scritto:

Dear all,
want to share my experience with SC-videocalls using jingle (Gmail) between two windows-labtops (nb 3081) calling Berlin-Vienna:

nice video (no delay in time)
audio ok (a bit jittering)
zRTP is working!

Congratulations! THIS IS REALLY GREAT! :slight_smile:

kind regards, MS

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#3

Well - I have an obstacle in the form of an router. (which Libnice and STUN are supposed to circumvent) This is the first version with Jingle activated - so I didn't expected it to work out of the box.

I hope, my observations might be of use for you to debug:

After trying to establish a call to someone across the Internet (NAT-NAT) wihout success, I concentated on the echo@test.collabora.co.uk echo test.

As long as "Use ICE" is deactivated, a call is established but no voice is coming back.

When I activate ICE - the call keeps ringing and doesn't get answered.

The config dialog for STUN asks for an account and password, which is strange as no other VoIP program ever did (probably for TURN??)

Anyway. I entered something for account/passwort there (as I had to) and configured stun.sipgate.net port 10000. Unfortunately to no avail. No voice coming back.

Given the fact that test.collabora.co.uk is not natted (for the RTPs at least) - it should work (even if my router was symmetric NAT which he isn't - he pretends to be VoIP friendly)

The echo Test works for Empathy (with Pidgin strangely not)

I hope that helps you to improve.

Conrad (impatiently waiting for the first free XMPP/Jingle client for Windows and Linux - thanks for your great work)

路路路

Datum: Tue, 16 Nov 2010 18:25:12 +0100
Von: Mr Smith <mr.smith476@googlemail.com>
An: dev@sip-communicator.dev.java.net
Betreff: [sip-comm-dev] jingle videocalls testing successful

Dear all,
want to share my experience with SC-videocalls using jingle (Gmail)
between two windows-labtops (nb 3081) calling Berlin-Vienna:

nice video (no delay in time)
audio ok (a bit jittering)
zRTP is working!

Congratulations! THIS IS REALLY GREAT! :slight_smile:

kind regards, MS

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#4

Hi All,

Yes i agree it is really great in terms of performances, but P2P
network condition is not always possible, and so there is no audio or
one way audio in many conditions, is there any thing that is
possible to do to discover P2P network condition or relay RTP in some
cases ? I know this is one of the limitation of Jingle to be a mass
production solution, but may be there is some solution for it
recently ?

The Google Talk team has found that ~8% of the time you need a relay.
The only solution is to deploy relays (e.g., TURN servers).

It will be great to have more details regarding mass deploying of the
solution

Volunteers are welcome. :slight_smile:

Another solution: Jingle Relay Nodes. You could run your own relay on
your home network, virtual private server, etc.

Peter

路路路

On 11/16/10 12:40 PM, Fabio Telme wrote:

--
Peter Saint-Andre
https://stpeter.im/


#5

Hi Peter,

Thanks for your explanation, if the 8% is the percentage of the times it needs relay should be accettable for mass deployment, but on the current release of jingle on the SC there should be some problem, i can estabilish a call only on local LAN, i use common residential ADSL in Italy (ALice) with NAT router with no firewall or restriction, and on this simple and common network enviroment i can run other P2P connection without any relay, expecially i'm testing Flash RTMP solution with Stratus 1.0 and Stratus 2.0. But with SC i can't get any audio from both party.
Is there anything i can check ?

Thanks
Fabio

路路路

Il giorno 16/nov/2010, alle ore 20.43, Peter Saint-Andre ha scritto:

On 11/16/10 12:40 PM, Fabio Telme wrote:

Hi All,

Yes i agree it is really great in terms of performances, but P2P
network condition is not always possible, and so there is no audio or
one way audio in many conditions, is there any thing that is
possible to do to discover P2P network condition or relay RTP in some
cases ? I know this is one of the limitation of Jingle to be a mass
production solution, but may be there is some solution for it
recently ?

The Google Talk team has found that ~8% of the time you need a relay.
The only solution is to deploy relays (e.g., TURN servers).

It will be great to have more details regarding mass deploying of the
solution

Volunteers are welcome. :slight_smile:

Another solution: Jingle Relay Nodes. You could run your own relay on
your home network, virtual private server, etc.

Peter

--
Peter Saint-Andre
https://stpeter.im/

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#6

Hi Fabio,

That is exactly why we are working on a jingle nodes based solution, where
have P2P traversal is trivial and also having Relay installed on XMPP
servers are also very easy, only requiring public IP.

For more information check: http://jinglenodes.org

Regards,
Thiago

路路路

On Tue, Nov 16, 2010 at 8:40 PM, Fabio Telme <fabio@telme.sg> wrote:

Hi All,

Yes i agree it is really great in terms of performances, but P2P network
condition is not always possible, and so there is no audio or one way audio
in many conditions, is there any thing that is possible to do to discover
P2P network condition or relay RTP in some cases ? I know this is one of the
limitation of Jingle to be a mass production solution, but may be there is
some solution for it recently ?

It will be great to have more details regarding mass deploying of the
solution

Regards
Fabio Galdi

Il giorno 16/nov/2010, alle ore 18.25, Mr Smith ha scritto:

> Dear all,
> want to share my experience with SC-videocalls using jingle (Gmail)
between two windows-labtops (nb 3081) calling Berlin-Vienna:
>
>
> nice video (no delay in time)
> audio ok (a bit jittering)
> zRTP is working!
>
> Congratulations! THIS IS REALLY GREAT! :slight_smile:
>
>
> kind regards, MS
>
>
> ---------------------------------------------------------------------
> To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
> For additional commands, e-mail: dev-help@sip-communicator.dev.java.net
>

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#7

Hey Conrad,

袧邪 17.11.10 01:23, Conrad Beckert 薪邪锌懈褋邪:

Well - I have an obstacle in the form of an router. (which Libnice
and STUN are supposed to circumvent) This is the first version with
Jingle activated - so I didn't expected it to work out of the box.

I hope, my observations might be of use for you to debug:

Yes, they are, thanks.

After trying to establish a call to someone across the Internet
(NAT-NAT) wihout success, I concentated on the
echo@test.collabora.co.uk echo test.

Yes indeed. ICE negotiation seems to be failing with them.

As long as "Use ICE" is deactivated, a call is established but no
voice is coming back.

Right ... that's not even supposed to work. We have activated ICE
support by default for all jabber accounts now and we'll probably do so
for TURN as well.

When I activate ICE - the call keeps ringing and doesn't get
answered.

We'll be checking this out.

The config dialog for STUN asks for an account and password, which is
strange as no other VoIP program ever did (probably for TURN??)

Indeed this is only necessary/used with TURN. We'll make sure we change
the GUI.

Anyway. I entered something for account/passwort there (as I had to)
and configured stun.sipgate.net port 10000. Unfortunately to no
avail. No voice coming back.

Given the fact that test.collabora.co.uk is not natted (for the RTPs
at least) - it should work (even if my router was symmetric NAT which
he isn't - he pretends to be VoIP friendly)

The echo Test works for Empathy (with Pidgin strangely not)

I hope that helps you to improve.

Yup. We're getting there :slight_smile:

Cheers,
Emil

Conrad (impatiently waiting for the first free XMPP/Jingle client for
Windows and Linux - thanks for your great work)

Datum: Tue, 16 Nov 2010 18:25:12 +0100 Von: Mr Smith
<mr.smith476@googlemail.com> An: dev@sip-communicator.dev.java.net
Betreff: [sip-comm-dev] jingle videocalls testing successful

Dear all, want to share my experience with SC-videocalls using
jingle (Gmail) between two windows-labtops (nb 3081) calling
Berlin-Vienna:

nice video (no delay in time) audio ok (a bit jittering) zRTP is
working!

Congratulations! THIS IS REALLY GREAT! :slight_smile:

kind regards, MS

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路路路

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#8

Hey Fabio,

I wanted to comment on this but couldn't do it earlier so apologies for
a late reply:

袧邪 16.11.10 20:40, Fabio Telme 薪邪锌懈褋邪:

is there any thing that is
possible to do to discover P2P network condition or relay RTP in some
cases ?

Yes, we are currently doing this with the ICE implementation from
http://ice4j.org. As I already mentioned all this is work in progress so
you are likely to have problems with it at this point. I expect most
issues to be resolve in the following week or two.

I know this is one of the limitation of Jingle to be a mass
production solution, but may be there is some solution for it
recently ?

Note that NAT traversal is hardly a Jingle limitation. Jingle has been
designed to use ICE from the start. Use of ICE also involves fallback to
TURN whenever relaying is absolutely necessary.

TURN is fully supported by ice4j.org and you can use
http://TurnServer.org for the server side.

As Thiago mentioned there's also Jingle Nodes. I believe Thiago is
currently working on integrating Jingle Nodes in SIP Communicator, so,
in other words, there's absolutely no reason to worry about NAT
traversal with Jingle.

Cheers,
Emil

It will be great to have more details regarding mass deploying of the
solution

Regards Fabio Galdi

Dear all, want to share my experience with SC-videocalls using
jingle (Gmail) between two windows-labtops (nb 3081) calling
Berlin-Vienna:

nice video (no delay in time) audio ok (a bit jittering) zRTP is
working!

Congratulations! THIS IS REALLY GREAT! :slight_smile:

kind regards, MS

---------------------------------------------------------------------

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dev-help@sip-communicator.dev.java.net

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路路路

Il giorno 16/nov/2010, alle ore 18.25, Mr Smith ha scritto:
For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#9

Hi Emil,

Thanks very much for the reply, just a simple question to use ICE it will not need any server side ? like for Stun or Turn ?

Thanks
Fabio

路路路

Il giorno 18/nov/2010, alle ore 13.10, Emil Ivov ha scritto:

Hey Fabio,

I wanted to comment on this but couldn't do it earlier so apologies for
a late reply:

袧邪 16.11.10 20:40, Fabio Telme 薪邪锌懈褋邪:

is there any thing that is
possible to do to discover P2P network condition or relay RTP in some
cases ?

Yes, we are currently doing this with the ICE implementation from
http://ice4j.org. As I already mentioned all this is work in progress so
you are likely to have problems with it at this point. I expect most
issues to be resolve in the following week or two.

I know this is one of the limitation of Jingle to be a mass
production solution, but may be there is some solution for it
recently ?

Note that NAT traversal is hardly a Jingle limitation. Jingle has been
designed to use ICE from the start. Use of ICE also involves fallback to
TURN whenever relaying is absolutely necessary.

TURN is fully supported by ice4j.org and you can use
http://TurnServer.org for the server side.

As Thiago mentioned there's also Jingle Nodes. I believe Thiago is
currently working on integrating Jingle Nodes in SIP Communicator, so,
in other words, there's absolutely no reason to worry about NAT
traversal with Jingle.

Cheers,
Emil

It will be great to have more details regarding mass deploying of the
solution

Regards Fabio Galdi

Il giorno 16/nov/2010, alle ore 18.25, Mr Smith ha scritto:

Dear all, want to share my experience with SC-videocalls using
jingle (Gmail) between two windows-labtops (nb 3081) calling
Berlin-Vienna:

nice video (no delay in time) audio ok (a bit jittering) zRTP is
working!

Congratulations! THIS IS REALLY GREAT! :slight_smile:

kind regards, MS

---------------------------------------------------------------------

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dev-help@sip-communicator.dev.java.net

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For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#10

Fabio,

ICE tries as much as possible to do NOT use Server, but in fact STUN, TURN
or Jingle Nodes will always be required in edge cases. Google Claims ~8%,
Personal experience shows around 20%.

If using TURN, you need to handle credentials and setup extra service.
If using Jingle Nodes it is plug and play and auto discoverable even when
outside your DNS range.

Thiago Rocha Camargo
barata7@gmail.com
http://jinglenodes.org
My profiles: [image: Twitter] <http://twitter.com/xmppjingle>
Contact me: [image: Google Talk/] barata7@gmail.com

路路路

On Thu, Nov 18, 2010 at 1:36 PM, Fabio Telme <fabio@telme.sg> wrote:

Hi Emil,

Thanks very much for the reply, just a simple question to use ICE it will
not need any server side ? like for Stun or Turn ?

Thanks
Fabio

Il giorno 18/nov/2010, alle ore 13.10, Emil Ivov ha scritto:

> Hey Fabio,
>
> I wanted to comment on this but couldn't do it earlier so apologies for
> a late reply:
>
> 袧邪 16.11.10 20:40, Fabio Telme 薪邪锌懈褋邪:
>> is there any thing that is
>> possible to do to discover P2P network condition or relay RTP in some
>> cases ?
>
> Yes, we are currently doing this with the ICE implementation from
> http://ice4j.org. As I already mentioned all this is work in progress so
> you are likely to have problems with it at this point. I expect most
> issues to be resolve in the following week or two.
>
>> I know this is one of the limitation of Jingle to be a mass
>> production solution, but may be there is some solution for it
>> recently ?
>
> Note that NAT traversal is hardly a Jingle limitation. Jingle has been
> designed to use ICE from the start. Use of ICE also involves fallback to
> TURN whenever relaying is absolutely necessary.
>
> TURN is fully supported by ice4j.org and you can use
> http://TurnServer.org for the server side.
>
> As Thiago mentioned there's also Jingle Nodes. I believe Thiago is
> currently working on integrating Jingle Nodes in SIP Communicator, so,
> in other words, there's absolutely no reason to worry about NAT
> traversal with Jingle.
>
> Cheers,
> Emil
>
>
>
>
>
>>
>> It will be great to have more details regarding mass deploying of the
>> solution
>>
>> Regards Fabio Galdi
>>
>>
>> Il giorno 16/nov/2010, alle ore 18.25, Mr Smith ha scritto:
>>
>>> Dear all, want to share my experience with SC-videocalls using
>>> jingle (Gmail) between two windows-labtops (nb 3081) calling
>>> Berlin-Vienna:
>>>
>>>
>>> nice video (no delay in time) audio ok (a bit jittering) zRTP is
>>> working!
>>>
>>> Congratulations! THIS IS REALLY GREAT! :slight_smile:
>>>
>>>
>>> kind regards, MS
>>>
>>>
>>> ---------------------------------------------------------------------
>>>
>>>
> To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
>>> For additional commands, e-mail:
>>> dev-help@sip-communicator.dev.java.net
>>>
>>
>>
>> ---------------------------------------------------------------------
>>
>>
> To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
>> For additional commands, e-mail:
>> dev-help@sip-communicator.dev.java.net
>>
>>
>
> --
> Emil Ivov, Ph.D. 67000 Strasbourg,
> Project Lead France
> SIP Communicator
> emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
> http://sip-communicator.org FAX: +33.1.77.62.47.31
>
>
> ---------------------------------------------------------------------
> To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
> For additional commands, e-mail: dev-help@sip-communicator.dev.java.net
>

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#11

袧邪 18.11.10 13:36, Fabio Telme 薪邪锌懈褋邪:

Hi Emil,

Thanks very much for the reply, just a simple question to use ICE it
will not need any server side ? like for Stun or Turn ?

Short answer is: In order to efficiently use ICE you need to have a TURN
enabled STUN server (or a Jingle Nodes relay) available on your network.

Slightly longer aswer is:

The ICE protocol itself does not require any server support. It simply
defines how clients exchange the addresses that they could potentially
receive media on. It also tells clients how they should verify which of
these addresses actually work. This is what ICE refers to as
"connectivity checks".

Having a STUN server available on your network would allow clients to
also advertise the address that they've been allocated by their NAT box
(ICE calls this a "server reflexive address").

Maintaining a TURN server would allow clients to use a relay in case
there's no way of establishing a more direct connection.

Cheers,
Emil

Thanks Fabio

Hey Fabio,

I wanted to comment on this but couldn't do it earlier so apologies
for a late reply:

袧邪 16.11.10 20:40, Fabio Telme 薪邪锌懈褋邪:

is there any thing that is possible to do to discover P2P network
condition or relay RTP in some cases ?

Yes, we are currently doing this with the ICE implementation from
http://ice4j.org. As I already mentioned all this is work in
progress so you are likely to have problems with it at this point.
I expect most issues to be resolve in the following week or two.

I know this is one of the limitation of Jingle to be a mass
production solution, but may be there is some solution for it
recently ?

Note that NAT traversal is hardly a Jingle limitation. Jingle has
been designed to use ICE from the start. Use of ICE also involves
fallback to TURN whenever relaying is absolutely necessary.

TURN is fully supported by ice4j.org and you can use
http://TurnServer.org for the server side.

As Thiago mentioned there's also Jingle Nodes. I believe Thiago is
currently working on integrating Jingle Nodes in SIP Communicator,
so, in other words, there's absolutely no reason to worry about
NAT traversal with Jingle.

Cheers, Emil

It will be great to have more details regarding mass deploying of
the solution

Regards Fabio Galdi

Dear all, want to share my experience with SC-videocalls using
jingle (Gmail) between two windows-labtops (nb 3081) calling
Berlin-Vienna:

nice video (no delay in time) audio ok (a bit jittering) zRTP
is working!

Congratulations! THIS IS REALLY GREAT! :slight_smile:

kind regards, MS

---------------------------------------------------------------------

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-- Emil Ivov, Ph.D. 67000
Strasbourg, Project Lead France
SIP Communicator emcho@sip-communicator.org
PHONE: +33.1.77.62.43.30 http://sip-communicator.org
FAX: +33.1.77.62.47.31

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路路路

Il giorno 18/nov/2010, alle ore 13.10, Emil Ivov ha scritto:

Il giorno 16/nov/2010, alle ore 18.25, Mr Smith ha scritto:

For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#12

Hi Thiago

Thanks for the explanation, so you aspect to have jingle nodes to be implemented in the SC?

In this case it will be great salution for it.

路路路

Inviato da iPhone

Il giorno 18/nov/2010, alle ore 13:59, thiagoc <barata7@gmail.com> ha scritto:

Fabio,

ICE tries as much as possible to do NOT use Server, but in fact STUN, TURN or Jingle Nodes will always be required in edge cases. Google Claims ~8%, Personal experience shows around 20%.

If using TURN, you need to handle credentials and setup extra service.
If using Jingle Nodes it is plug and play and auto discoverable even when outside your DNS range.

Thiago Rocha Camargo
barata7@gmail.com
http://jinglenodes.org
My profiles:
Contact me: barata7@gmail.com

On Thu, Nov 18, 2010 at 1:36 PM, Fabio Telme <fabio@telme.sg> wrote:
Hi Emil,

Thanks very much for the reply, just a simple question to use ICE it will not need any server side ? like for Stun or Turn ?

Thanks
Fabio

Il giorno 18/nov/2010, alle ore 13.10, Emil Ivov ha scritto:

> Hey Fabio,
>
> I wanted to comment on this but couldn't do it earlier so apologies for
> a late reply:
>
> 袧邪 16.11.10 20:40, Fabio Telme 薪邪锌懈褋邪:
>> is there any thing that is
>> possible to do to discover P2P network condition or relay RTP in some
>> cases ?
>
> Yes, we are currently doing this with the ICE implementation from
> http://ice4j.org. As I already mentioned all this is work in progress so
> you are likely to have problems with it at this point. I expect most
> issues to be resolve in the following week or two.
>
>> I know this is one of the limitation of Jingle to be a mass
>> production solution, but may be there is some solution for it
>> recently ?
>
> Note that NAT traversal is hardly a Jingle limitation. Jingle has been
> designed to use ICE from the start. Use of ICE also involves fallback to
> TURN whenever relaying is absolutely necessary.
>
> TURN is fully supported by ice4j.org and you can use
> http://TurnServer.org for the server side.
>
> As Thiago mentioned there's also Jingle Nodes. I believe Thiago is
> currently working on integrating Jingle Nodes in SIP Communicator, so,
> in other words, there's absolutely no reason to worry about NAT
> traversal with Jingle.
>
> Cheers,
> Emil
>
>
>
>
>
>>
>> It will be great to have more details regarding mass deploying of the
>> solution
>>
>> Regards Fabio Galdi
>>
>>
>> Il giorno 16/nov/2010, alle ore 18.25, Mr Smith ha scritto:
>>
>>> Dear all, want to share my experience with SC-videocalls using
>>> jingle (Gmail) between two windows-labtops (nb 3081) calling
>>> Berlin-Vienna:
>>>
>>>
>>> nice video (no delay in time) audio ok (a bit jittering) zRTP is
>>> working!
>>>
>>> Congratulations! THIS IS REALLY GREAT! :slight_smile:
>>>
>>>
>>> kind regards, MS
>>>
>>>
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>>>
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>
> --
> Emil Ivov, Ph.D. 67000 Strasbourg,
> Project Lead France
> SIP Communicator
> emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
> http://sip-communicator.org FAX: +33.1.77.62.47.31
>
>
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