Open Source Sip: [OpenSIPStack] An enhanced HTTP Config ...

11 Replies Last post: Dec 9, 2007 8:34 PM by Guest

Guest

Dec 2, 2007 10:42 PM

[OpenSIPStack] An enhanced HTTP Config Module for OpenSBC

Hi Everyone,

As I have hinted in the past, there is a plan to provide a new level of
administrative interface for OpenSBC. As most of you might have
discovered by now, OpenSBC is very easy to install and requires
virtually no configuration for you to be able to run and use it. This
is all because of a built-in HTTP server that allows for OpenSBC to be
configured remotely. However, since version 1.1.4 and with the
introduction of more advanced features like SIP Trunking, the built-in
HTTP Config Pages is out-growing its simplicity. We are now in a
point where we need to decide what technology to use to bring the
configuration modules to the next level. We need to seriously
consider the following criteria in choosing the solution.

1. It should be very easy to install and package
2. Built-in access to back-end databases preferably Postgress
3. Must be able to host Dynamic HTML Content
4. Other criteria that might be useful are support for XML-RPC and SOAP

Some of the basic choices that I pulled out straight from my thoughts are

1. A separate web interface easily installable as standard Apache Web
application. Choices are PHP, JSP Ruby with SQL back-end
2. Stand-alone Python HTTP Server just like the trac project. See
http://trac.edgewall.org/
3. Enhance the HTTP Admin to support Fast-CGI extension to allow for
dynamic HTTP content to be hosted straight by OpenSBC

I would want to hear from everyone if you are leaning towards a certain
approach or would want another approach altogether.

Joegen

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Guest

1. Dec 3, 2007 3:14 AM

in response to: Guest

Re: [OpenSIPStack] An enhanced HTTP Config Module for OpenSBC

Standalone Python server like Trac and/or Twisted.web+woven sounds good to me

Just my 2c though

-----Original Message-----
From: opensipstack-devel-bounces@lists.sourceforge.net opensipstack-devel-bounces@lists.sourceforge.net On Behalf Of Joegen E. Baclor
Sent: Montag, 03. Dezember 2007 04:43
To: opensipstack-devel@lists.sourceforge.net; opensipstack-osbcdevel@lists.sourceforge.net
Subject: OpenSIPStack An enhanced HTTP Config Module for OpenSBC

Hi Everyone,

As I have hinted in the past, there is a plan to provide a new level of
administrative interface for OpenSBC. As most of you might have
discovered by now, OpenSBC is very easy to install and requires
virtually no configuration for you to be able to run and use it. This
is all because of a built-in HTTP server that allows for OpenSBC to be
configured remotely. However, since version 1.1.4 and with the
introduction of more advanced features like SIP Trunking, the built-in
HTTP Config Pages is out-growing its simplicity. We are now in a
point where we need to decide what technology to use to bring the
configuration modules to the next level. We need to seriously
consider the following criteria in choosing the solution.

1. It should be very easy to install and package
2. Built-in access to back-end databases preferably Postgress
3. Must be able to host Dynamic HTML Content
4. Other criteria that might be useful are support for XML-RPC and SOAP

Guest

2. Dec 3, 2007 9:15 AM

in response to: Guest

Re: [OpenSIPStack] An enhanced HTTP Config Module for OpenSBC

Hi

Support for MySQL would be nice alternative and perhaps a plug-in port for
etherreal/Firewire type analyzer to read the Logs could be very useful.

Warren Kreckler

Original Message

From: "Joegen E. Baclor"
To: ;

Sent: Sunday, December 02, 2007 9:42 PM
Subject: OpenSIPStack An enhanced HTTP Config Module for OpenSBC

Guest

3. Dec 5, 2007 2:46 AM

in response to: Guest

Re: [OpenSIPStack] An enhanced HTTP Config Module for OpenSBC

To sum up. One in the list is in favor of a python approach and another
in favor of a built-in fast CGI approach. Everyone seems unanimous in
having SQL server support. Just this morning, Ryan Colobong pointed me
to http://www.cherrypy.org/ which is another python based solution to
incorporate dynamic web pages to opensbc. There seem to be a lot of
resources we could use in a python based solution. However, fast-cgi
support in OpenSBC would carry on the tradition of simplicity in
installation, it will take a lot longer just to develop the module that
enabled fast-cgi support not to mention that HTTP is not really the turf
OpenSIPStack. At this point I am leaning towards a python based
solution. For those who are willing to pitch in your ideas, now is
the time to speak up.

Joegen

Joegen E. Baclor wrote:

Guest

4. Dec 5, 2007 10:17 PM

in response to: Guest

Re: [OpenSIPStack] B2BUA how to route

Hi

Almost have this puppy working.

Sipx and opensbc generally well understood.

Problem:

When OSBC receives INVITE from sipX => ITSP,
OSBC route the INVITE back to sipX.

We have two rules in the B2Bua route
~~sip:202.100.2.23:5060~~ sip: 202.100.2.23 this goes to our ITSP
~~sip:sipx.sip.net:5060~~ sip:sipx.sip.net this point to our sipx

the missing rule/route?

Where do you put the rule and what should the rule say to route INVITE out
to our ITSP?

Warren Kreckler

Guest

5. Dec 7, 2007 1:08 AM

in response to: Guest

Re: [OpenSIPStack] B2BUA how to route

You need to use the SIP Trunking capability of OpenSBC for this. Do
you need to authenticate calls with your ITSP?

sales@ER wrote:

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Guest

6. Dec 7, 2007 11:52 AM

in response to: Guest

Re: [OpenSIPStack] B2BUA how to route

Yes They call it peer to peer. By that they meam

1. Via Headers: ITSP has stated that they can accept only 1 Via
statement in an INVITE. As background, each device will add a Via statement
to the INVITE to if it has processed the INVITE. Only the last or top entry
is really of interest to the party that next handles the INVITE. In order
for ITSP to accept the INVITE of an outbound call, OpenSBC will
need to strip off all previous Via statements from the INVITE and add its'
own. I have not found any capability to remove the previously inserted Via
statements.

2. Lock IP Address and port to first sender: This option comes into play
when a call has been answered either by a person or system component (i.e.
Auto Attendant) and a transfer is attempted. When the transferred call is
answered by a new phone or component, it will negotiate use of a new RTP
port for the media stream. Some service providers, ITSP included,
do not allow the RTP port to change once the initial call is established.
They do this to protect against the "hijacking" of a call by Hackers. Since
the media is flowing through a SBC, the SBC then needs to manage which ports
are used to exchange media (voice). If the original port is not utilized
for the media back to the carrier, the PSTN will not hear any audio once the
call is transferred. I do not see this capability with OpenSBC.

3. Calling ID: SIPxchange utilizes the From: element to provide the
Calling ID (DID). It normally inserts the userID in the user part of the

From URI. ITSP uses the INVITE element

Remote-Party-ID to determine the Calling ID. This is not an element created
by Sipx. The SBC will need to extract the user part of the From URI and
create a Remote-Party-ID. I did not see this capability with OpenSBC.
Without this, the called party on the PSTN will either see "Private
Caller"or "Anonymous" on their phone instead of the DID.

Warren Kreckler

Original Message

From: "Joegen E. Baclor"
To:
Cc:
Sent: Friday, December 07, 2007 12:08 AM
Subject: Re: OpenSIPStack B2BUA how to route

You need to use the SIP Trunking capability of OpenSBC for this. Do
you need to authenticate calls with your ITSP?

sales@ER wrote:

sipx

the missing rule/route?

Where do you put the rule and what should the rule say to route INVITE

out

to our ITSP?

Warren Kreckler

Guest

7. Dec 8, 2007 2:11 PM

in response to: Guest

Re: [OpenSIPStack] SIP Trunk how to route

Hi

Is there a doc for SIP Trunk? When and why to use it and relationship to
B2Bua etc...?

Warren Kreckler

Guest

8. Dec 8, 2007 6:35 PM

in response to: Guest

Re: [OpenSIPStack] (no subject)

Hi

Can some one help me understand the relationship between B2Bua, Sip Trunk
and Upper Registration Routes. With some examples

lets assume:
SBC 200.10.10.5
Sip Server 200.10.10.10
Lan 200.10.10.0/24
ip Phone 200.10.10.101
ITSP 100.10.10.10

Is there a round robin hand-off relationship or conflick between B2Bua and
Sip Trunk.
Im getting a error "OpenSBC transport Status" in the Web Admin tool

In the Sip Trunk i have 100.10.10.10 == ITSP

SIP TRUNK:
200.10.10.10 :5060 error Address Already in use **** Failed ******

Well there ever be workig How To doc?

Gaurav Kheterpal sent in a pcap to the list
but it was removed.

Warren Kreckler

Guest

9. Dec 8, 2007 6:40 PM

in response to: Guest

Re: [OpenSIPStack] Sip Trunk, B2Bua and Upper RR

Guest

10. Dec 9, 2007 8:21 PM

in response to: Guest

Re: [OpenSIPStack] B2BUA how to route

inline...

sales@ER wrote:

Yes They call it peer to peer. By that they meam

What version are you using? There was a bug introduced when we got
back from sipIT 21 due to the changes made there that had the vias not
getting stripped. Please use the latest CVS. OpenSBC should be
stripping the via before the B2BUA sends the INVITE out to the UAS.

2. Lock IP Address and port to first sender: This option comes into play
when a call has been answered either by a person or system component (i.e.
Auto Attendant) and a transfer is attempted. When the transferred call is
answered by a new phone or component, it will negotiate use of a new RTP
port for the media stream. Some service providers, ITSP included,
do not allow the RTP port to change once the initial call is established.
They do this to protect against the "hijacking" of a call by Hackers. Since
the media is flowing through a SBC, the SBC then needs to manage which ports
are used to exchange media (voice). If the original port is not utilized
for the media back to the carrier, the PSTN will not hear any audio once the
call is transferred. I do not see this capability with OpenSBC.

In media proxy mode (Always Proxy Media = true), OpenSBC does not change
the port of RTP even during reInvites.

3. Calling ID: SIPxchange utilizes the From: element to provide the
Calling ID (DID). It normally inserts the userID in the user part of the

From URI. ITSP uses the INVITE element

Can you send a sample of this from header that is rewritten by sipX?

Warren Kreckler

Original Message

From: "Joegen E. Baclor"
To:
Cc:
Sent: Friday, December 07, 2007 12:08 AM
Subject: Re: OpenSIPStack B2BUA how to route

You need to use the SIP Trunking capability of OpenSBC for this. Do
you need to authenticate calls with your ITSP?

sales@ER wrote:

sipx

the missing rule/route?

Where do you put the rule and what should the rule say to route INVITE

out

to our ITSP?

Warren Kreckler

Guest

11. Dec 9, 2007 8:34 PM

in response to: Guest

Re: [OpenSIPStack] SIP Trunk how to route

sales@ER wrote:

Hi

Is there a doc for SIP Trunk? When and why to use it and relationship to
B2Bua etc...?

Warren Kreckler

Unfortunately not. SIP Trunking is very new in the arsenal and would
be hard to use without a more advanced configuration module. For now,
there is a temporary provision to test it using the config pages via
XML. I have attached the annoucement i've sent last October 17, 2007

To tell OpenSBC that the call should be processed by the SIP Trunk your B2BUARoute entry shouold include a sip-trunk paramameter.

Example:
sip:1212* sip:mytrunkprovider.com <http://mytrunkprovider.com>;sip-trunk=true

Hi,

SIP Trunking is not prime time yet but you may already try it using the
latest CVS copy of OpenSBC/OpenSIPStack. To Enable Trunking, you must
provide an XML configuration in "SIP Trunk Config". Below is a template
XML config. In this sample config, sip:win32.opensipstack.org is
assumed to be the internal domain of OpenSBC while
sip:opteron.opensipstack.org is the domain of your SIP Provider.

SIPTrunk

trunk-name: This is the unique name OpenSBC will use to identify you

SIP Trunk

route-set: This is the DNS resolvable domain or IP address of your

trunk provider

sip-domain: This is the SIP Domain used as the host part of the To

and From URIs

expires: Global expire interval for trunk registrations in seconds

Trunk-Accounts

account - An instance of a virtual UA that will register to the Trunk

Provider domain

user-name - The user part of the From-URI
auth-user-name - User name used for Authorization and Authentication
auth-password - Password used for Authorization and Authentication
inbound-route - URI specifying the identity of the UA in the internal

domain

expires - If set, this will be the expires used when the virtual UA

registers to the Trunk Provider

Transient-Accounts - Transient accounts are similar to normal
Trunk-Account in terms of the parameters. The only difference is that
they are also meant to be shared (in round robin fashion) by calls which
are not defined in the normal trunk-accounts. This is normally used if
you have a few accounts with a Trunk Provider and is meant to be shared
by all your external users.

------------------------START OF XML
CONFIG----------------------------------

Joegen

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Dec 2, 2007 10:42 PM

[OpenSIPStack] An enhanced HTTP Config Module for OpenSBC

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