SIP Trunking

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SIP Trunking

Matthew S. Crocker

Hello,

 I'm brand new to OpenSIPS, just going through the make process now.  

 I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch.  Where should I look for Documentation/Examples of a working config?

Here is my scenario:

OpenSIPS has two interfaces,  private & public.  
VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing)

End user has an Asterisk server on a private lan behind their firewall (NAT)

I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall.  It then need to rewrite the SIP message and send it to the Gateway.  The Gateway would see the messages coming from the internal IP of the OpenSIPS server.  Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway.

Any helpful hints on where to look?

-Matt


--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
P: 413-746-2760


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Re: SIP Trunking

Steven C. Blair

I did this once before. I would suggest dividing the config file into two pieces. The first handled outbound to the PSTN, the second inbound from the PSTN. This allows you to rewrite header fields based on your requirements or those of your ITSP in a fairly straightforward way

-steve

-----Original Message-----
From: [hidden email] [mailto:[hidden email]] On Behalf Of Matthew S. Crocker
Sent: Thursday, August 20, 2009 1:54 PM
To: [hidden email]
Subject: [OpenSIPS-Users] SIP Trunking


Hello,

 I'm brand new to OpenSIPS, just going through the make process now.  

 I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch.  Where should I look for Documentation/Examples of a working config?

Here is my scenario:

OpenSIPS has two interfaces,  private & public.  
VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing)

End user has an Asterisk server on a private lan behind their firewall (NAT)

I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall.  It then need to rewrite the SIP message and send it to the Gateway.  The Gateway would see the messages coming from the internal IP of the OpenSIPS server.  Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway.

Any helpful hints on where to look?

-Matt


--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
P: 413-746-2760


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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Re: SIP Trunking

Alex Balashov
In reply to this post by Matthew S. Crocker
Matthew,

Look for the mediaproxy module.

That said, do be aware that a proxy is, by definition, not like an SBC.
  SBCs have many other capabilities a proxy does not;  a proxy is a
relatively "thin" interoperation layer.

Perhaps the recently introduced b2bua module is brought to bear on that
somewhat, but classically, OpenSIPS is a proxy.

-- Alex

Matthew S. Crocker wrote:

> Hello,
>
>  I'm brand new to OpenSIPS, just going through the make process now.  
>
>  I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch.  Where should I look for Documentation/Examples of a working config?
>
> Here is my scenario:
>
> OpenSIPS has two interfaces,  private & public.  
> VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing)
>
> End user has an Asterisk server on a private lan behind their firewall (NAT)
>
> I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall.  It then need to rewrite the SIP message and send it to the Gateway.  The Gateway would see the messages coming from the internal IP of the OpenSIPS server.  Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway.
>
> Any helpful hints on where to look?
>
> -Matt
>
>


--
Alex Balashov - Principal
Evariste Systems
Web     : http://www.evaristesys.com/
Tel     : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: SIP Trunking

Matthew S. Crocker

I understand that OpenSIPS is not a full blown SBC (I can't afford an ACMEPacket).  Will it perform the functions to proxy the SIP & RTP streams (via mediaproxy) between my end users and my internal gateway?

At some point I plan on increasing the use of openSIPS to handle registration, presence, routing, etc.

-Matt

----- "Alex Balashov" <[hidden email]> wrote:

> From: "Alex Balashov" <[hidden email]>
> To: "OpenSIPS users mailling list" <[hidden email]>
> Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern
> Subject: Re: [OpenSIPS-Users] SIP Trunking
>
> Matthew,
>
> Look for the mediaproxy module.
>
> That said, do be aware that a proxy is, by definition, not like an
> SBC.
>   SBCs have many other capabilities a proxy does not;  a proxy is a
> relatively "thin" interoperation layer.
>
> Perhaps the recently introduced b2bua module is brought to bear on
> that
> somewhat, but classically, OpenSIPS is a proxy.
>
> -- Alex
>
> Matthew S. Crocker wrote:
>
> > Hello,
> >
> >  I'm brand new to OpenSIPS, just going through the make process now.
>  
> >
> >  I need to configure OpenSIPS to act like a SBC for some SIP trunks
> coming off a VoIP switch.  Where should I look for
> Documentation/Examples of a working config?
> >
> > Here is my scenario:
> >
> > OpenSIPS has two interfaces,  private & public.  
> > VoIP Gateway is on private LAN with no gateway configured (it can
> only talk to local machines, no routing)
> >
> > End user has an Asterisk server on a private lan behind their
> firewall (NAT)
> >
> > I need to configure OpenSIPS to listen for SIP messages on :5060
> from the end user firewall.  It then need to rewrite the SIP message
> and send it to the Gateway.  The Gateway would see the messages coming
> from the internal IP of the OpenSIPS server.  Once all of the SIP
> messages get processed I then need the OpenSIPS server to proxy the
> RTP streams (plan on using mediaproxy) between the Asterisk server and
> VoIP Gateway.
> >
> > Any helpful hints on where to look?
> >
> > -Matt
> >
> >
>
>
> --
> Alex Balashov - Principal
> Evariste Systems
> Web     : http://www.evaristesys.com/
> Tel     : (+1) (678) 954-0670
> Direct  : (+1) (678) 954-0671
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
P: 413-746-2760


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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: SIP Trunking

Jeff Pyle
Matthew,

While I'm no Mediaproxy expert, I have seen many conversations on this list
where Mediaproxy is described as a part of a far-end NAT solution.  It was
not designed to have a private IP attached to it.  For that, you most likely
will want to look at the rtpproxy application.

It sounds like you are constructing a local ALG to connect private and
public networks.  You don't necessarily need a full-blown Acme for that.
I've had great luck with Edgewater Networks' "Edgemarc" devices, for
example.  That's just one.  There are many.


- Jeff



On 8/20/09 2:49 PM, "Matthew S. Crocker" <[hidden email]> wrote:

>
> I understand that OpenSIPS is not a full blown SBC (I can't afford an
> ACMEPacket).  Will it perform the functions to proxy the SIP & RTP streams
> (via mediaproxy) between my end users and my internal gateway?
>
> At some point I plan on increasing the use of openSIPS to handle registration,
> presence, routing, etc.
>
> -Matt
>
> ----- "Alex Balashov" <[hidden email]> wrote:
>
>> From: "Alex Balashov" <[hidden email]>
>> To: "OpenSIPS users mailling list" <[hidden email]>
>> Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada Eastern
>> Subject: Re: [OpenSIPS-Users] SIP Trunking
>>
>> Matthew,
>>
>> Look for the mediaproxy module.
>>
>> That said, do be aware that a proxy is, by definition, not like an
>> SBC.
>>   SBCs have many other capabilities a proxy does not;  a proxy is a
>> relatively "thin" interoperation layer.
>>
>> Perhaps the recently introduced b2bua module is brought to bear on
>> that
>> somewhat, but classically, OpenSIPS is a proxy.
>>
>> -- Alex
>>
>> Matthew S. Crocker wrote:
>>
>>> Hello,
>>>
>>>  I'm brand new to OpenSIPS, just going through the make process now.
>>  
>>>
>>>  I need to configure OpenSIPS to act like a SBC for some SIP trunks
>> coming off a VoIP switch.  Where should I look for
>> Documentation/Examples of a working config?
>>>
>>> Here is my scenario:
>>>
>>> OpenSIPS has two interfaces,  private & public.
>>> VoIP Gateway is on private LAN with no gateway configured (it can
>> only talk to local machines, no routing)
>>>
>>> End user has an Asterisk server on a private lan behind their
>> firewall (NAT)
>>>
>>> I need to configure OpenSIPS to listen for SIP messages on :5060
>> from the end user firewall.  It then need to rewrite the SIP message
>> and send it to the Gateway.  The Gateway would see the messages coming
>> from the internal IP of the OpenSIPS server.  Once all of the SIP
>> messages get processed I then need the OpenSIPS server to proxy the
>> RTP streams (plan on using mediaproxy) between the Asterisk server and
>> VoIP Gateway.
>>>
>>> Any helpful hints on where to look?
>>>
>>> -Matt
>>>
>>>
>>
>>
>> --
>> Alex Balashov - Principal
>> Evariste Systems
>> Web     : http://www.evaristesys.com/
>> Tel     : (+1) (678) 954-0670
>> Direct  : (+1) (678) 954-0671
>>
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: SIP Trunking

Darren Sessions-2
In reply to this post by Matthew S. Crocker
Media Proxy will work with public internet addresses but, and this is  
just my understanding, was not built for use with private IPs let  
alone bridging from one subnet to another.

If your keeping the gateways on a private network is for security  
purposes, you may consider giving them public ips on the same subnet  
as your opensips and mediaproxy setup, but not specifying a default  
gateway.

Essentially, this would allow the media proxy to do its job relaying  
the audio, while still preventing 99% of any unwanted traffic to your  
gateways. Couple that will firehol or some other cool iptables app (or  
manually configure it if you like) and you'd be sitting pretty secure  
I would think.

Really depends on what you've designed (and why).

- Darren


On Aug 20, 2009, at 12:49 PM, Matthew S. Crocker wrote:

>
> I understand that OpenSIPS is not a full blown SBC (I can't afford  
> an ACMEPacket).  Will it perform the functions to proxy the SIP &  
> RTP streams (via mediaproxy) between my end users and my internal  
> gateway?
>
> At some point I plan on increasing the use of openSIPS to handle  
> registration, presence, routing, etc.
>
> -Matt
>
> ----- "Alex Balashov" <[hidden email]> wrote:
>
>> From: "Alex Balashov" <[hidden email]>
>> To: "OpenSIPS users mailling list" <[hidden email]>
>> Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada  
>> Eastern
>> Subject: Re: [OpenSIPS-Users] SIP Trunking
>>
>> Matthew,
>>
>> Look for the mediaproxy module.
>>
>> That said, do be aware that a proxy is, by definition, not like an
>> SBC.
>>  SBCs have many other capabilities a proxy does not;  a proxy is a
>> relatively "thin" interoperation layer.
>>
>> Perhaps the recently introduced b2bua module is brought to bear on
>> that
>> somewhat, but classically, OpenSIPS is a proxy.
>>
>> -- Alex
>>
>> Matthew S. Crocker wrote:
>>
>>> Hello,
>>>
>>> I'm brand new to OpenSIPS, just going through the make process now.
>>
>>>
>>> I need to configure OpenSIPS to act like a SBC for some SIP trunks
>> coming off a VoIP switch.  Where should I look for
>> Documentation/Examples of a working config?
>>>
>>> Here is my scenario:
>>>
>>> OpenSIPS has two interfaces,  private & public.
>>> VoIP Gateway is on private LAN with no gateway configured (it can
>> only talk to local machines, no routing)
>>>
>>> End user has an Asterisk server on a private lan behind their
>> firewall (NAT)
>>>
>>> I need to configure OpenSIPS to listen for SIP messages on :5060
>> from the end user firewall.  It then need to rewrite the SIP message
>> and send it to the Gateway.  The Gateway would see the messages  
>> coming
>> from the internal IP of the OpenSIPS server.  Once all of the SIP
>> messages get processed I then need the OpenSIPS server to proxy the
>> RTP streams (plan on using mediaproxy) between the Asterisk server  
>> and
>> VoIP Gateway.
>>>
>>> Any helpful hints on where to look?
>>>
>>> -Matt
>>>
>>>
>>
>>
>> --
>> Alex Balashov - Principal
>> Evariste Systems
>> Web     : http://www.evaristesys.com/
>> Tel     : (+1) (678) 954-0670
>> Direct  : (+1) (678) 954-0671
>>
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> --
> Matthew S. Crocker
> President
> Crocker Communications, Inc.
> PO BOX 710
> Greenfield, MA 01302-0710
> http://www.crocker.com
> P: 413-746-2760
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: SIP Trunking

Ghaith ALKAYYEM
It's possible also to use RTPproxy, because it's designed to work in
bridging mode where it can forward the traffic from the internal network
towards external one.

Regards

On Thu, 2009-08-20 at 12:55 -0600, Darren Sessions wrote:

> Media Proxy will work with public internet addresses but, and this is  
> just my understanding, was not built for use with private IPs let  
> alone bridging from one subnet to another.
>
> If your keeping the gateways on a private network is for security  
> purposes, you may consider giving them public ips on the same subnet  
> as your opensips and mediaproxy setup, but not specifying a default  
> gateway.
>
> Essentially, this would allow the media proxy to do its job relaying  
> the audio, while still preventing 99% of any unwanted traffic to your  
> gateways. Couple that will firehol or some other cool iptables app (or  
> manually configure it if you like) and you'd be sitting pretty secure  
> I would think.
>
> Really depends on what you've designed (and why).
>
> - Darren
>
>
> On Aug 20, 2009, at 12:49 PM, Matthew S. Crocker wrote:
>
> >
> > I understand that OpenSIPS is not a full blown SBC (I can't afford  
> > an ACMEPacket).  Will it perform the functions to proxy the SIP &  
> > RTP streams (via mediaproxy) between my end users and my internal  
> > gateway?
> >
> > At some point I plan on increasing the use of openSIPS to handle  
> > registration, presence, routing, etc.
> >
> > -Matt
> >
> > ----- "Alex Balashov" <[hidden email]> wrote:
> >
> >> From: "Alex Balashov" <[hidden email]>
> >> To: "OpenSIPS users mailling list" <[hidden email]>
> >> Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada  
> >> Eastern
> >> Subject: Re: [OpenSIPS-Users] SIP Trunking
> >>
> >> Matthew,
> >>
> >> Look for the mediaproxy module.
> >>
> >> That said, do be aware that a proxy is, by definition, not like an
> >> SBC.
> >>  SBCs have many other capabilities a proxy does not;  a proxy is a
> >> relatively "thin" interoperation layer.
> >>
> >> Perhaps the recently introduced b2bua module is brought to bear on
> >> that
> >> somewhat, but classically, OpenSIPS is a proxy.
> >>
> >> -- Alex
> >>
> >> Matthew S. Crocker wrote:
> >>
> >>> Hello,
> >>>
> >>> I'm brand new to OpenSIPS, just going through the make process now.
> >>
> >>>
> >>> I need to configure OpenSIPS to act like a SBC for some SIP trunks
> >> coming off a VoIP switch.  Where should I look for
> >> Documentation/Examples of a working config?
> >>>
> >>> Here is my scenario:
> >>>
> >>> OpenSIPS has two interfaces,  private & public.
> >>> VoIP Gateway is on private LAN with no gateway configured (it can
> >> only talk to local machines, no routing)
> >>>
> >>> End user has an Asterisk server on a private lan behind their
> >> firewall (NAT)
> >>>
> >>> I need to configure OpenSIPS to listen for SIP messages on :5060
> >> from the end user firewall.  It then need to rewrite the SIP message
> >> and send it to the Gateway.  The Gateway would see the messages  
> >> coming
> >> from the internal IP of the OpenSIPS server.  Once all of the SIP
> >> messages get processed I then need the OpenSIPS server to proxy the
> >> RTP streams (plan on using mediaproxy) between the Asterisk server  
> >> and
> >> VoIP Gateway.
> >>>
> >>> Any helpful hints on where to look?
> >>>
> >>> -Matt
> >>>
> >>>
> >>
> >>
> >> --
> >> Alex Balashov - Principal
> >> Evariste Systems
> >> Web     : http://www.evaristesys.com/
> >> Tel     : (+1) (678) 954-0670
> >> Direct  : (+1) (678) 954-0671
> >>
> >> _______________________________________________
> >> Users mailing list
> >> [hidden email]
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> > --
> > Matthew S. Crocker
> > President
> > Crocker Communications, Inc.
> > PO BOX 710
> > Greenfield, MA 01302-0710
> > http://www.crocker.com
> > P: 413-746-2760
> >
> >
> > _______________________________________________
> > Users mailing list
> > [hidden email]
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


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Re: SIP Trunking

Matthew S. Crocker
In reply to this post by Jeff Pyle

Can mediaproxy glue two RTP streams together (CallerA to CallerB)?
Can mediaproxy glue two RTP streams together from different interfaces/IPs (eth0 & eth1) ?

If so then it should be able to glue two calls together between public IP (eth0) and private IP (eth1).
If the two RTP streams have to be on the same interface for mediaproxy to work then I would expect to run into issues.

EndUser <-> (eth0) MediaProxy (eth1) <-> SIP Gateway


----- "Jeff Pyle" <[hidden email]> wrote:

> From: "Jeff Pyle" <[hidden email]>
> To: "OpenSIPS users mailling list" <[hidden email]>
> Sent: Thursday, August 20, 2009 2:52:35 PM GMT -05:00 US/Canada Eastern
> Subject: Re: [OpenSIPS-Users] SIP Trunking
>
> Matthew,
>
> While I'm no Mediaproxy expert, I have seen many conversations on this
> list
> where Mediaproxy is described as a part of a far-end NAT solution.  It
> was
> not designed to have a private IP attached to it.  For that, you most
> likely
> will want to look at the rtpproxy application.
>
> It sounds like you are constructing a local ALG to connect private
> and
> public networks.  You don't necessarily need a full-blown Acme for
> that.
> I've had great luck with Edgewater Networks' "Edgemarc" devices, for
> example.  That's just one.  There are many.
>
>
> - Jeff
>
>
>
> On 8/20/09 2:49 PM, "Matthew S. Crocker" <[hidden email]>
> wrote:
>
> >
> > I understand that OpenSIPS is not a full blown SBC (I can't afford
> an
> > ACMEPacket).  Will it perform the functions to proxy the SIP & RTP
> streams
> > (via mediaproxy) between my end users and my internal gateway?
> >
> > At some point I plan on increasing the use of openSIPS to handle
> registration,
> > presence, routing, etc.
> >
> > -Matt
> >
> > ----- "Alex Balashov" <[hidden email]> wrote:
> >
> >> From: "Alex Balashov" <[hidden email]>
> >> To: "OpenSIPS users mailling list" <[hidden email]>
> >> Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada
> Eastern
> >> Subject: Re: [OpenSIPS-Users] SIP Trunking
> >>
> >> Matthew,
> >>
> >> Look for the mediaproxy module.
> >>
> >> That said, do be aware that a proxy is, by definition, not like an
> >> SBC.
> >>   SBCs have many other capabilities a proxy does not;  a proxy is
> a
> >> relatively "thin" interoperation layer.
> >>
> >> Perhaps the recently introduced b2bua module is brought to bear on
> >> that
> >> somewhat, but classically, OpenSIPS is a proxy.
> >>
> >> -- Alex
> >>
> >> Matthew S. Crocker wrote:
> >>
> >>> Hello,
> >>>
> >>>  I'm brand new to OpenSIPS, just going through the make process
> now.
> >>  
> >>>
> >>>  I need to configure OpenSIPS to act like a SBC for some SIP
> trunks
> >> coming off a VoIP switch.  Where should I look for
> >> Documentation/Examples of a working config?
> >>>
> >>> Here is my scenario:
> >>>
> >>> OpenSIPS has two interfaces,  private & public.
> >>> VoIP Gateway is on private LAN with no gateway configured (it can
> >> only talk to local machines, no routing)
> >>>
> >>> End user has an Asterisk server on a private lan behind their
> >> firewall (NAT)
> >>>
> >>> I need to configure OpenSIPS to listen for SIP messages on :5060
> >> from the end user firewall.  It then need to rewrite the SIP
> message
> >> and send it to the Gateway.  The Gateway would see the messages
> coming
> >> from the internal IP of the OpenSIPS server.  Once all of the SIP
> >> messages get processed I then need the OpenSIPS server to proxy
> the
> >> RTP streams (plan on using mediaproxy) between the Asterisk server
> and
> >> VoIP Gateway.
> >>>
> >>> Any helpful hints on where to look?
> >>>
> >>> -Matt
> >>>
> >>>
> >>
> >>
> >> --
> >> Alex Balashov - Principal
> >> Evariste Systems
> >> Web     : http://www.evaristesys.com/
> >> Tel     : (+1) (678) 954-0670
> >> Direct  : (+1) (678) 954-0671
> >>
> >> _______________________________________________
> >> Users mailing list
> >> [hidden email]
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
P: 413-746-2760


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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: SIP Trunking

Alex Balashov
Matthew S. Crocker wrote:

> Can mediaproxy glue two RTP streams together (CallerA to CallerB)?
> Can mediaproxy glue two RTP streams together from different interfaces/IPs (eth0 & eth1) ?

Yes.


--
Alex Balashov - Principal
Evariste Systems
Web     : http://www.evaristesys.com/
Tel     : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: SIP Trunking

Ghaith ALKAYYEM
In reply to this post by Matthew S. Crocker
Hello,

I tried to use mediaproxy, it includes two softwares (dispatcher &
relay), I tried a lot to run more than one relay on the same server in
order to bind them to different interfaces. But unfortunately this
didn't work and I think it's not possible.
I recommend using RTPProxy which is designed to work in bridging mode
between two networks and you can run multiple instance of RTPProxy on
the same server.

Regards.


On Thu, 2009-08-20 at 15:48 -0400, Matthew S. Crocker wrote:

> Can mediaproxy glue two RTP streams together (CallerA to CallerB)?
> Can mediaproxy glue two RTP streams together from different interfaces/IPs (eth0 & eth1) ?
>
> If so then it should be able to glue two calls together between public IP (eth0) and private IP (eth1).
> If the two RTP streams have to be on the same interface for mediaproxy to work then I would expect to run into issues.
>
> EndUser <-> (eth0) MediaProxy (eth1) <-> SIP Gateway
>
>
> ----- "Jeff Pyle" <[hidden email]> wrote:
>
> > From: "Jeff Pyle" <[hidden email]>
> > To: "OpenSIPS users mailling list" <[hidden email]>
> > Sent: Thursday, August 20, 2009 2:52:35 PM GMT -05:00 US/Canada Eastern
> > Subject: Re: [OpenSIPS-Users] SIP Trunking
> >
> > Matthew,
> >
> > While I'm no Mediaproxy expert, I have seen many conversations on this
> > list
> > where Mediaproxy is described as a part of a far-end NAT solution.  It
> > was
> > not designed to have a private IP attached to it.  For that, you most
> > likely
> > will want to look at the rtpproxy application.
> >
> > It sounds like you are constructing a local ALG to connect private
> > and
> > public networks.  You don't necessarily need a full-blown Acme for
> > that.
> > I've had great luck with Edgewater Networks' "Edgemarc" devices, for
> > example.  That's just one.  There are many.
> >
> >
> > - Jeff
> >
> >
> >
> > On 8/20/09 2:49 PM, "Matthew S. Crocker" <[hidden email]>
> > wrote:
> >
> > >
> > > I understand that OpenSIPS is not a full blown SBC (I can't afford
> > an
> > > ACMEPacket).  Will it perform the functions to proxy the SIP & RTP
> > streams
> > > (via mediaproxy) between my end users and my internal gateway?
> > >
> > > At some point I plan on increasing the use of openSIPS to handle
> > registration,
> > > presence, routing, etc.
> > >
> > > -Matt
> > >
> > > ----- "Alex Balashov" <[hidden email]> wrote:
> > >
> > >> From: "Alex Balashov" <[hidden email]>
> > >> To: "OpenSIPS users mailling list" <[hidden email]>
> > >> Sent: Thursday, August 20, 2009 1:58:48 PM GMT -05:00 US/Canada
> > Eastern
> > >> Subject: Re: [OpenSIPS-Users] SIP Trunking
> > >>
> > >> Matthew,
> > >>
> > >> Look for the mediaproxy module.
> > >>
> > >> That said, do be aware that a proxy is, by definition, not like an
> > >> SBC.
> > >>   SBCs have many other capabilities a proxy does not;  a proxy is
> > a
> > >> relatively "thin" interoperation layer.
> > >>
> > >> Perhaps the recently introduced b2bua module is brought to bear on
> > >> that
> > >> somewhat, but classically, OpenSIPS is a proxy.
> > >>
> > >> -- Alex
> > >>
> > >> Matthew S. Crocker wrote:
> > >>
> > >>> Hello,
> > >>>
> > >>>  I'm brand new to OpenSIPS, just going through the make process
> > now.
> > >>  
> > >>>
> > >>>  I need to configure OpenSIPS to act like a SBC for some SIP
> > trunks
> > >> coming off a VoIP switch.  Where should I look for
> > >> Documentation/Examples of a working config?
> > >>>
> > >>> Here is my scenario:
> > >>>
> > >>> OpenSIPS has two interfaces,  private & public.
> > >>> VoIP Gateway is on private LAN with no gateway configured (it can
> > >> only talk to local machines, no routing)
> > >>>
> > >>> End user has an Asterisk server on a private lan behind their
> > >> firewall (NAT)
> > >>>
> > >>> I need to configure OpenSIPS to listen for SIP messages on :5060
> > >> from the end user firewall.  It then need to rewrite the SIP
> > message
> > >> and send it to the Gateway.  The Gateway would see the messages
> > coming
> > >> from the internal IP of the OpenSIPS server.  Once all of the SIP
> > >> messages get processed I then need the OpenSIPS server to proxy
> > the
> > >> RTP streams (plan on using mediaproxy) between the Asterisk server
> > and
> > >> VoIP Gateway.
> > >>>
> > >>> Any helpful hints on where to look?
> > >>>
> > >>> -Matt
> > >>>
> > >>>
> > >>
> > >>
> > >> --
> > >> Alex Balashov - Principal
> > >> Evariste Systems
> > >> Web     : http://www.evaristesys.com/
> > >> Tel     : (+1) (678) 954-0670
> > >> Direct  : (+1) (678) 954-0671
> > >>
> > >> _______________________________________________
> > >> Users mailing list
> > >> [hidden email]
> > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> > _______________________________________________
> > Users mailing list
> > [hidden email]
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


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Re: SIP Trunking

Dan Pascu
In reply to this post by Jeff Pyle
On 20 Aug 2009, at 21:52, Jeff Pyle wrote:

> Matthew,
>
> While I'm no Mediaproxy expert, I have seen many conversations on  
> this list
> where Mediaproxy is described as a part of a far-end NAT solution.  
> It was
> not designed to have a private IP attached to it.

Just to eliminate any misunderstanding, mediaproxy can work just fine  
with a private IP address. Why we recommend to always use it with a  
public IP address, is because in that case it is able to automatically  
deal with the deadlock that occurs when multiple media relays are  
chained. When private IPs are used, you are responsible for solving  
the deadlock yourself and considering that you have no idea when  
another media relay may be chained into the media path, nor who is  
operating that media relay and under what assumptions, you are most  
likely unable to solve the deadlock issue when using private IPs.  
Unless you are in a strictly controlled environment where you can  
employ a specific protocol that would prevent media relays to be  
chained, you are better off using a public IP. If all this is not an  
issue for you, then feel free to use a private IP address.

--
Dan




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Re: SIP Trunking

Bogdan-Andrei Iancu
In reply to this post by Matthew S. Crocker
Hi Matthew,

There 2 things when comes bridging:

1) signalling part - selecting the proper outbound interface (private or
public)
    a) this can be automatically done by opensips (based on the
destination IP) if you enable the mhomed parameter in core ; this is
simple by not so efficient

    b) you can do it manually, by selecting from script the correct
interface - see the force_send_socket() function

2) media part
     a) rtpproxy - when enabling RTPproxy (at request and reply time)
you can explicitly select which interface to use (see the e and i flags
- http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id271362)


Best regards,
Bogdan

Matthew S. Crocker wrote:

> Hello,
>
>  I'm brand new to OpenSIPS, just going through the make process now.  
>
>  I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch.  Where should I look for Documentation/Examples of a working config?
>
> Here is my scenario:
>
> OpenSIPS has two interfaces,  private & public.  
> VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing)
>
> End user has an Asterisk server on a private lan behind their firewall (NAT)
>
> I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall.  It then need to rewrite the SIP message and send it to the Gateway.  The Gateway would see the messages coming from the internal IP of the OpenSIPS server.  Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway.
>
> Any helpful hints on where to look?
>
> -Matt
>
>
>  


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Re: SIP Trunking

Ghaith ALKAYYEM
Hi,

Is it possible also to make bridging dependent on a variable value by
passing a variable as a parameter to force_send_socket() as following:

$var(a) = "x.x.x.x:xx";
force_send_socket("$var(a)");

because the above configuration gave me an error but when I used the
variable in xlog function it was okay:
xlog("$var(a)");

I might do some code modification in this regard.

Regards.

On Mon, 2009-08-24 at 18:03 +0300, Bogdan-Andrei Iancu wrote:

> Hi Matthew,
>
> There 2 things when comes bridging:
>
> 1) signalling part - selecting the proper outbound interface (private or
> public)
>     a) this can be automatically done by opensips (based on the
> destination IP) if you enable the mhomed parameter in core ; this is
> simple by not so efficient
>
>     b) you can do it manually, by selecting from script the correct
> interface - see the force_send_socket() function
>
> 2) media part
>      a) rtpproxy - when enabling RTPproxy (at request and reply time)
> you can explicitly select which interface to use (see the e and i flags
> - http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id271362)
>
>
> Best regards,
> Bogdan
>
> Matthew S. Crocker wrote:
> > Hello,
> >
> >  I'm brand new to OpenSIPS, just going through the make process now.  
> >
> >  I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch.  Where should I look for Documentation/Examples of a working config?
> >
> > Here is my scenario:
> >
> > OpenSIPS has two interfaces,  private & public.  
> > VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing)
> >
> > End user has an Asterisk server on a private lan behind their firewall (NAT)
> >
> > I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall.  It then need to rewrite the SIP message and send it to the Gateway.  The Gateway would see the messages coming from the internal IP of the OpenSIPS server.  Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway.
> >
> > Any helpful hints on where to look?
> >
> > -Matt
> >
> >
> >  
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


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Re: SIP Trunking

Bogdan-Andrei Iancu
Hi Ghaith,

Force_send_socket() does not accept a variable,but you can use instead
the $fs (force socket) var which does accept variables :

$var(a) = "x.x.x.x:xx";
$fs = $var(a) ;

Regards,
Bogdan



Ghaith ALKAYYEM wrote:

> Hi,
>
> Is it possible also to make bridging dependent on a variable value by
> passing a variable as a parameter to force_send_socket() as following:
>
> $var(a) = "x.x.x.x:xx";
> force_send_socket("$var(a)");
>
> because the above configuration gave me an error but when I used the
> variable in xlog function it was okay:
> xlog("$var(a)");
>
> I might do some code modification in this regard.
>
> Regards.
>
> On Mon, 2009-08-24 at 18:03 +0300, Bogdan-Andrei Iancu wrote:
>  
>> Hi Matthew,
>>
>> There 2 things when comes bridging:
>>
>> 1) signalling part - selecting the proper outbound interface (private or
>> public)
>>     a) this can be automatically done by opensips (based on the
>> destination IP) if you enable the mhomed parameter in core ; this is
>> simple by not so efficient
>>
>>     b) you can do it manually, by selecting from script the correct
>> interface - see the force_send_socket() function
>>
>> 2) media part
>>      a) rtpproxy - when enabling RTPproxy (at request and reply time)
>> you can explicitly select which interface to use (see the e and i flags
>> - http://www.opensips.org/html/docs/modules/1.5.x/nathelper.html#id271362)
>>
>>
>> Best regards,
>> Bogdan
>>
>> Matthew S. Crocker wrote:
>>    
>>> Hello,
>>>
>>>  I'm brand new to OpenSIPS, just going through the make process now.  
>>>
>>>  I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch.  Where should I look for Documentation/Examples of a working config?
>>>
>>> Here is my scenario:
>>>
>>> OpenSIPS has two interfaces,  private & public.  
>>> VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing)
>>>
>>> End user has an Asterisk server on a private lan behind their firewall (NAT)
>>>
>>> I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall.  It then need to rewrite the SIP message and send it to the Gateway.  The Gateway would see the messages coming from the internal IP of the OpenSIPS server.  Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway.
>>>
>>> Any helpful hints on where to look?
>>>
>>> -Matt
>>>
>>>
>>>  
>>>      
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>    
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>  


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Re: SIP Trunking

ahmeddd
In reply to this post by Matthew S. Crocker
Hallo!

Can any one help me in configuring SIP trunk in a local notwork ! I have two IP phones and I have already installed opensips in my VM, and I want to route calls between those two IP phones.

thnx
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Re: SIP Trunking

Bogdan-Andrei Iancu-2
Hi,

The OpenSIPS default script (or any generated via "make menuconfig" or
"osipsconfig" for residential scenario) will do the trick for you -
allowing multiple phones to register and call one each other.

Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com

OpenSIPS Bootcamp 2017, Houston, US
   http://opensips.org/training/OpenSIPS_Bootcamp_2017.html

On 07/17/2017 04:24 PM, ahmeddd wrote:

> Hallo!
>
> Can any one help me in configuring SIP trunk in a local notwork ! I have two
> IP phones and I have already installed opensips in my VM, and I want to
> route calls between those two IP phones.
>
> thnx
>
>
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-Trunking-tp3480673p7608019.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: SIP Trunking

ahmeddd
Hi!
yes it's clear, but I mean how to enable the trunk between two different IP
phones which should be registered to my opensips proxy !! should I do
somethig special to register them ! or should I specify the domain :( VM1 :
x.x.13.87 and VM2 x.x.13.82 ) and for my opensips server (x.x.15.18) ! or
it's the same domain as x.x.15.18 !!

thank you .



--
View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-Trunking-tp3480673p7608027.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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Re: SIP Trunking

ahmeddd
In reply to this post by Bogdan-Andrei Iancu-2
Hi!
yes it's clear, but I mean how to enable the trunk between two different IP phones which should be registered to my opensips proxy !! should I do somethig special to register them ! or should I specify the domain :( VM1 : x.x.13.87 and VM2 x.x.13.82 ) and for my opensips server (x.x.15.18) ! or it's the same domain as x.x.15.18 !!
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Re: SIP Trunking

ahmeddd
Can any one help me in this fact !!
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