[jitsi-dev] Enquiry about the possibility to configure Jitsi for VSAT network.


#1

Dear Jitsi developers group,
In the frame of the deployment of the low-cost SatADSL VSAT network in sub-saharan Africa, I would like to use jitsi as the basic VoIP solution, but this would work only if the following mandatory configuration can be managed by Jitsi softphone:
- G729A with 5 frames/IP packet (ptime=50ms) in order to get RTP IP packet of 90bytes. Alternatively, g723.1 could be used with sample size per IP packet = 60ms (=pTime), which makes IP packet size of 80 or 88 bytes (at 5.3 or 6.4 kbps accordingly).- protocols and jitter buffers must be able to accommodate an additional delay of 700ms due to satellite round trip delay: for the SIP protocol, it means RFC 3261 T1 value (RTT estimate) must be set = 2 sec; RFC 3261 T2 value (maximum re-transmit interval for non-INVITE requests and INVITE responses) must be set = 8 sec, and RFC 3261 T4 value (maximum duration a message remains in the network) must be set = 8 sec.- to have the possibility to adjust DSCP value to 0x33 in hexadecimal or 51 in decimal for RTP (voice) traffic and to 0x2E in hexadecimal or 46 in decimal for SIP (signaling) traffic. Alternatively, a single DSCP value of 51 for both SIP and RTP should also work.
It must be understood that if this can be achieved, this will bring the possibility for all sub-saharan people to access to low-cost international communications through VSAT !
All suggestion are welcome.
All testing can be managed very quickly by myself.
Thanks in advance for taking this request into consideration.
Don't hesitate to recontact me for any further information
Michel DOTHEYmichel.dothey@satadsl.net


#2

Hey Michel,

Dear Jitsi developers group,

In the frame of the deployment of the low-cost SatADSL VSAT network in
sub-saharan Africa, I would like to use jitsi as the basic VoIP
solution, but this would work only if the following mandatory
configuration can be managed by Jitsi softphone:

- G729A with 5 frames/IP packet (ptime=50ms) in order to get RTP IP
packet of 90bytes. Alternatively, g723.1 could be used with sample size
per IP packet = 60ms (=pTime), which makes IP packet size of 80 or 88
bytes (at 5.3 or 6.4 kbps accordingly).

We do support G.729 Annex C with libjitsi and Jitsi. We don't enable it
by default since you need to acquire the necessary license in order to
use it.

That said, we support various other codecs that have both a better
quality and a comparable or lower bandwidth. SILK is one such choice.
We've also just received a contribution with Opus which definitely beats
all the others. You can use both of these free of charge.

To put it differently: you need to have very very good reason in order
to insist on G.729.

- protocols and jitter buffers must be able
to accommodate an additional delay of 700ms due to satellite round trip
delay: for the SIP protocol, it means RFC 3261 T1 value (RTT estimate)
must be set = 2 sec; RFC 3261 T2 value (maximum re-transmit interval for
non-INVITE requests and INVITE responses) must be set = 8 sec, and RFC
3261 T4 value (maximum duration a message remains in the network) must
be set = 8 sec.

All these are configurable in JSIP and we can easily export those
configuration options through Jitsi too.

- to have the possibility to adjust DSCP value to 0x33 in hexadecimal or
51 in decimal for RTP (voice) traffic and to 0x2E in hexadecimal or 46
in decimal for SIP (signaling) traffic. Alternatively, a single DSCP
value of 51 for both SIP and RTP should also work.

You can set DSCP values for both of these in Jitsi (and for XMPP too).

It must be understood that if this can be achieved, this will bring the
possibility for all sub-saharan people to access to low-cost
international communications through VSAT !

We are glad to hear this and wish you good luck with your project!

Cheers,
Emil

···

On 11.09.12, 14:36, Michel DOTHEY wrote:

All suggestion are welcome.

All testing can be managed very quickly by myself.

Thanks in advance for taking this request into consideration.

Don't hesitate to recontact me for any further information

Michel DOTHEY
michel.dothey@satadsl.net

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


#3

Hi Emil,
It seems from your signature that you are living in France, so I propose to continue en français....
Merci pour ta réponse extrêmement encourageante.
1. Au sujet du choix du codec: en fait ce choix résulte de plusieurs contraintes, dont la plus importante provient des concepteurs techniques du système VSAT DVB-S2 que nous utilisons et qui a priori nous imposent pour le canat temps réel VoIP des trames RTP de max 90 octets toutes les 50ms ou de 80 ou 88 octets toutes les 60ms. Ceci en pratique correspond aux codec suivants:- G729A avec un ptime de 50ms (=5 trames / packet IP) ce qui donne des paquet IP de 90 octets- G723.1avec un ptime de 60ms ce qui donne des paquet IP de 80 ou 88 octets (à 5.3kbps ou 6.4kbps respectivement).Ceci dit, je pense que d'autres vocodeurs (à fort taux de compression vu l'étroitesse du canal : 60kbps) plus performants pourraient être utilisés, mais il faut qu'ils puissent fournir des trames du type de celles décrites ci-dessus.Mais dans ce cas, on se heurterait à un deuxième problème qui est la compatibilité avec le monde extérieur, dont essentiellement avec les opérateurs VoIP pour les appels vers le réseau téléphonique mondial, qui n'utilisent qu'un nombre restreint de vocodeurs.En effet, pour offrir les maximum de possibilités à nos utilisateurs, idéalement il faudrait que ceux-ci puissent:- communiquer gratuitement entre membres d'un même réseau (fermé)(par exemple en adressant directement l'IP fixe ou autre) et via un maximum de réseaux (ouverts) existants du type MSN, GoogleTalk, Facebook, etc.- se connecter (via protocole SIP) au réseau téléphonique mondial via un opérateur VoIP moyennant l'achat d'un numéro de téléphone n'importe où dans le monde.
Je crains donc que nous n'ayons donc guère le choix et que nous soyons bel et bien obligés d'utiliser le G729A qui est le plus répendu.
2. Par ailleurs, je suis ravi de savoir que vous pouvez imposer les priorités DSCP aux valeurs imposées tant pour les protocoles SIP que XMPP.(Ils semblerait toutefois que par défaut Windows7 remettrait les DSCP à 0, sauf modification de certains registres. As-tu déjà entendu parler de ce problème??)
Ceci dit, nous n'avons jusqu'à présent su tester que le protocole SIP sur notre système. Il serait pour nous extrêmement intéressant de savoir si le protocole XMPP pourrait également être utilisé, car cela nous ouvrirait les portes à Jabber, Facebook et Google Talk entre autres. Ceci toutefois devrait être testé car le délai supplémentaire imposé par les systèmes satellitaires nécessitent souvent des ajustement dans le timing des différents protocoles.
Au vu de ce qui précède, pourrais-tu me dire comment procéder pour pour démarrer des tests le plus rapidement possible. L'idéal pour nous serait de disposer d'un prototype avec les différents paramètres configurables afin de pouvoir procéder à leur ajustement. Ensuite, une fois ceux-ci définis, une version de production pourrait être déployée auprès de nos utilisateurs.
Pourrais-tu me confirmer si cette approche correspond à votre manière de procéder, et si des frais sont à prévoir (et si oui à combien s'élèvent-ils) ?
Encore merci.
Au plaisir de te lire,
Bien à toi,
Michel DOTHEYwww.satadsl.netrue Royale 1821000 BruxellesBelgique+32 495 53 06 12michel.dothey@satadsl.net
NB: If you prefer to continue in English, just let me know, and I will translate the above in French.

···

Date: Sat, 15 Sep 2012 12:04:17 +0300
From: emcho@jitsi.org
To: dev@jitsi.java.net
CC: mdothey@hotmail.com; fulvio.sansone@satadsl.net
Subject: Re: [jitsi-dev] Enquiry about the possibility to configure Jitsi for VSAT network. >
Hey Michel,

On 11.09.12, 14:36, Michel DOTHEY wrote:
> Dear Jitsi developers group,
>
> In the frame of the deployment of the low-cost SatADSL VSAT network in
> sub-saharan Africa, I would like to use jitsi as the basic VoIP
> solution, but this would work only if the following mandatory
> configuration can be managed by Jitsi softphone:
>
> - G729A with 5 frames/IP packet (ptime=50ms) in order to get RTP IP
> packet of 90bytes. Alternatively, g723.1 could be used with sample size
> per IP packet = 60ms (=pTime), which makes IP packet size of 80 or 88
> bytes (at 5.3 or 6.4 kbps accordingly).

We do support G.729 Annex C with libjitsi and Jitsi. We don't enable it
by default since you need to acquire the necessary license in order to
use it.

That said, we support various other codecs that have both a better
quality and a comparable or lower bandwidth. SILK is one such choice.
We've also just received a contribution with Opus which definitely beats
all the others. You can use both of these free of charge.

To put it differently: you need to have very very good reason in order
to insist on G.729.

> - protocols and jitter buffers must be able
> to accommodate an additional delay of 700ms due to satellite round trip
> delay: for the SIP protocol, it means RFC 3261 T1 value (RTT estimate)
> must be set = 2 sec; RFC 3261 T2 value (maximum re-transmit interval for
> non-INVITE requests and INVITE responses) must be set = 8 sec, and RFC
> 3261 T4 value (maximum duration a message remains in the network) must
> be set = 8 sec.

All these are configurable in JSIP and we can easily export those
configuration options through Jitsi too.

> - to have the possibility to adjust DSCP value to 0x33 in hexadecimal or
> 51 in decimal for RTP (voice) traffic and to 0x2E in hexadecimal or 46
> in decimal for SIP (signaling) traffic. Alternatively, a single DSCP
> value of 51 for both SIP and RTP should also work.

You can set DSCP values for both of these in Jitsi (and for XMPP too).

> It must be understood that if this can be achieved, this will bring the
> possibility for all sub-saharan people to access to low-cost
> international communications through VSAT !

We are glad to hear this and wish you good luck with your project!

Cheers,
Emil

> All suggestion are welcome.
>
> All testing can be managed very quickly by myself.
>
> Thanks in advance for taking this request into consideration.
>
> Don't hesitate to recontact me for any further information
>
> Michel DOTHEY
> michel.dothey@satadsl.net
>
>

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