Open Source Sip: Re: [OpenSIPStack] Dialog events ...

44 Replies Last post: Mar 25, 2008 7:15 AM by Guest Go to original post 1 2 3 Previous Next

Guest

30. Dec 11, 2007 9:39 PM

in response to: Guest

Re: [OpenSIPStack] Playing a Recorded File

Whit,

There is currently no provision to play wav files through the softphone
interface. If you want to dig deeper, you need to look at the media
server implementation. Ilian is in the best position to give you more
details how this can be done. I am sorry I can't be of further
assistance to you on this.

Joegen

Whit Thiele wrote:

Hey Guys,

Is it possible to play a recorded file (like a .wav file) from the ATLSIP
Softphone so that it plays and can be heard by both the caller and the
callee at the same time?

I've seen a lot of references in the code to Play File, but not really sure
how it works (ie. VoiceFileChannel::PlayFile) Any tips would be great...

Thanks,

Whit

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Guest

31. Dec 12, 2007 3:52 AM

in response to: Guest

Re: [OpenSIPStack] Playing a Recorded File

Hi Whit,

Whit Thiele wrote:

Hey Guys,

Is it possible to play a recorded file (like a .wav file) from the ATLSIP

As Joegen has said, this functionality is not readily available in ATLSIP.

Softphone so that it plays and can be heard by both the caller and the
callee at the same time?

You will need a mixer for this condition.

The way I see it this can probably be implemented via a filter. Refer to
OpalPCSSConnection::OnPatchMediaStream(..) and check how
OpalSilenceDetector and OpalEchoCanceler filters handle it. Another
filter class is needed to handle the mixing of the streams with a wav
file (PWavFile).

I can probably implement this myself but it will take a while because I
have other higher priority projects.

Regards,
Ilian

I've seen a lot of references in the code to Play File, but not really sure
how it works (ie. VoiceFileChannel::PlayFile) Any tips would be great...

Thanks,

Whit

Guest

32. Dec 23, 2007 12:13 PM

in response to: Guest

Re: [OpenSIPStack] [OpenSBC] Sip Trunck Howto

Hi ehernaez

Thank you for the pdf link.

My guess is that there are many definitions of What/Who/Where is a Sip Trunk
provider and what they expect in terms of param's.

I have several Sip Trunk providers. How can i use this form for multi's?
And they seem to range from smart to stupid again in terms what they expect
and whether if they discard useless param or see the extra stuff as attacks.

While seaching for the proper form of the xml i saw deferences in the
examples. For instance one contains the brackets others
don't. Is the required?

One Sip Truck provider uses a peering arrangement. They setup an 'one of a
kind port' for us and expect two param's the DID and an ipaddr simliar to
an email address.

Can i safely remove parameters i don't need and that the ST provider see as
junk, unexpected and a threat?

r

Original Message

From: "ehernaez"
To:
Sent: Friday, December 14, 2007 10:20 AM
Subject: Re: OpenSIPStack OpenSBC Sip Trunck Howto

Please see here:

http://www.opensourcesip.org:8080/clearspacex/docs/DOC-1040

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Guest

33. Dec 26, 2007 8:35 AM

in response to: Guest

Re: [OpenSIPStack] [OpenSBC] Sip Trunck Howto

I am having trouble following your example. Could you please post a more specific example of what your ITSP is expecting?

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Guest

34. Dec 26, 2007 11:59 AM

in response to: Guest

Re: [OpenSIPStack] [OpenSBC] Sip Trunck Howto

Hi ehernaez

They are a carrier using Sonus box.

They make the distinction between Peering and Access SBC.

They want to establish a Peer to Peer connection. They defined this
arrangement as Carrier to Carrier. These two SBC connection are linked via
the Internet and will be use for no other purpose.

They have provide us both DID's and access to an unique ipaddr. They expect
from us one of those DID's and that unique ipaddr in the From: header when
we send sip traffic, which is part of the security they use to identify
9094441024@x.x.x.241

ipaddr/SBC/peer x.x.x.190 <==> DID@x.x.x.241

On our sipX box we have two (2) Sip Trunk Gateways. Each pointing to one of
those OSBC box's.

One pointing to OSBC, which will be used for Peering edge and the other OBSC
used for Access edge,phone registration etc...

Warren Kreckler

Original Message

From: "ehernaez"
To:
Sent: Wednesday, December 26, 2007 7:35 AM
Subject: Re: OpenSIPStack OpenSBC Sip Trunck Howto

I am having trouble following your example. Could you please post a more

specific example of what your ITSP is expecting?

Guest

35. Dec 29, 2007 7:47 AM

in response to: Guest

Re: [OpenSIPStack] [OpenSBC] Two (2) OSBC on net segment

Hi

Is it possible to pass phone registration from one OSBC through too another
OSBC.

I am experimenting with two OSBC's on the same network segment with an
sipXecs in between. One setup as Peering edge and the other as Access edge.

Warren Kreckler

Guest

36. Dec 29, 2007 8:00 AM

in response to: Guest

Re: [OpenSIPStack] [OpenSBC] Two (2) OSBC on net segment

Hi Warren,

If you mean mirroring of registrations via none signaling means then the
answer is no. Upper registration may let the registration pass but I
seriously doubt this is what you mean.

Joegen

sales@ER wrote:

Guest

37. Dec 29, 2007 9:17 AM

in response to: Guest

Re: [OpenSIPStack] [OpenSBC] Two (2) OSBC on net segment

Hi Joegen

xxxxxxxxxxxxxxxxxxx
Here is a re-cap Re: OpenSBC Peering Edge or Access Edge SBC's
Warren,

Yes, you are correct in concluding that you would need two OSBC
instances in the scenario you laid out. The instance that you use for
SIP trunking to face your ITSP should not be the same instance that you
are using for NAT traversal of SIP UAs using upper registration to SIPX.

In fact, it may even be necessary for you to have 3 instances of OSBC if
you need to support DID providers that do not comply with SIP REFER.
As you may know, calls transferred from the SIPX auto-attendant are
referred to the relevant UA. Since many ITSP do not honor the REFER
message, we have found it necessary to run a dedicated OSBC instance as
a media anchor to shield the ITSP from the referred calls.

HTH
xxxxxxxxxxxxxxxxxxxxxxxxxx

In the above scenario ITSB Peering OSBC(po) <==> sipX <==> Access OSBC(ao).
You can think of po as a pipe/bridge/trunk or two hardwired SBC's via the
Internet.

Without the po active on network I am able to register phones to the ao and
accept phone calls but an not able to make phone calls. In this scenario
calls to the ao are sent back to the sipX. The solution is to add a second
OSBC used as Peering Edge. So now the trick is to setup OSBC as a Peering
edge SBC. In my simplistic thinking because the ITSP on the Peering Edge is
both looking for a registered phone on inbound calls and looking for From;
DID@PeeringEdge to allow sip passage outbound but on the Access side
outbound look like this From DID@AccessEdge .

SipFoundary list is also tackeling this issue but is un-aware that a second
or a third OSBC may be required. If what HTH said is true how do we bridge
the OSBC Peering and Access Edges

Warren Kreckler

Original Message

From: "Joegen E. Baclor"
To:
Sent: Saturday, December 29, 2007 7:00 AM
Subject: Re: OpenSIPStack OpenSBC Two (2) OSBC on net segment

Hi

Is it possible to pass phone registration from one OSBC through too

another

OSBC.

I am experimenting with two OSBC's on the same network segment with an
sipXecs in between. One setup as Peering edge and the other as Access

edge.

Warren Kreckler

Guest

38. Dec 29, 2007 10:58 AM

in response to: Guest

Re: [OpenSIPStack] [OpenSBC] Two (2) OSBC on net segment

Hi Warren,

Here is the call flow that you can use to bridge the 2 OSBC instances:
For an inbound call from PSTN: ITSP -> Peering OSBC -> Access OSBC
-> SIPXFor an outbound cal to the PSTN: UA -> Access OSBC -> SIPX -> Peering
OSBC -> ITSP
*inbound calls must be routed through the access OSBC in order to reach
endpoints that are registered thru the access OSBC
HTH

Guest

39. Dec 29, 2007 11:37 AM

in response to: Guest

Re: [OpenSIPStack] [OpenSBC] Two (2) OSBC on net segment

Hi ehernaez

ITSP <==> Peering OSBC <==> Access OSBC <==> sipX This requires two (2)
opensbc's
UA <==> Access OSBC <==> sipX <==> Peering OSBC <==> ITSP. This requires
two (2) additional OSBC

For a grand total of four (4) OSBC's. That's 5 servers altogether.

Another way of looking at this is using factoring:

ITSP <==> Peering OSBC <==> Access OSBC <==> sipX

II

II
UA <==> Access OSBC <==> sipX <==>
Peering OSBC <==> ITSP

or this

ITSP <==> Peering OSBC <==> Access OSBC <==> sipX <==> UA <==> Access OSBC
<==> sipX <==> Peering OSBC <==> ITSP

This is beginning to make me laugh out loud. In other words there is no way
of using UPPerRegistration and Relay routes to eliminate 2 of the OSBC's

It's beginning to look like the sipX people should incorporate openSBC as a
SIP TRUNK Gateway for both Peering and Access Edge SBC's. After all openSBC
is 100% RFC compliant. My god 4 OSBC's servers...

Warrem Kreckler.

Original Message

From: "ehernaez"
To:
Sent: Saturday, December 29, 2007 9:58 AM
Subject: Re: OpenSIPStack OpenSBC Two (2) OSBC on net segment

Hi Warren,

Here is the call flow that you can use to bridge the 2 OSBC instances:
For an inbound call from PSTN: ITSP -> Peering OSBC -> Access

OSBC

-> SIPXFor an outbound cal to the PSTN: UA -> Access OSBC ->

SIPX -> Peering

OSBC -> ITSP
*inbound calls must be routed through the access OSBC in order to reach
endpoints that are registered thru the access OSBC
HTH

Guest

40. Dec 30, 2007 2:41 PM

in response to: Guest

Re: [OpenSIPStack] [OpenSBC] Two (2) OSBC on net segment

Hi ehernaez

Using your first example inbound call from PSTN: ITSP -> Peering
OSBC -> Access OSBC -> SIPX
What mode would you set?
What route (b2bua, relay) would you recomend setting on the Peering OSBC to
reach the Acceess OSBC or is a matter of Trusted Domain and setting either
in not required?

Warren Kreckler

Original Message

From: "ehernaez"
To:
Sent: Saturday, December 29, 2007 9:58 AM
Subject: Re: OpenSIPStack OpenSBC Two (2) OSBC on net segment

Hi Warren,

Here is the call flow that you can use to bridge the 2 OSBC instances:
For an inbound call from PSTN: ITSP -> Peering OSBC -> Access

OSBC

-> SIPXFor an outbound cal to the PSTN: UA -> Access OSBC ->

SIPX -> Peering

OSBC -> ITSP
*inbound calls must be routed through the access OSBC in order to reach
endpoints that are registered thru the access OSBC
HTH

Guest

41. Dec 30, 2007 3:06 PM

in response to: Guest

Re: [OpenSIPStack] [OpenSBC] Two (2) OSBC on net segment

Hi ehernaiz

We experimennted all night and this is the last one we tried but failed.
Please take a look and point out the mistake?

This is your recommendation for outbound call to the PSTN:
UA -> Access OSBC -> SIPX -> Peering OSBC -> ITSP

We are using the same two OSBC as for inbound calls. This first half seems
to work fine
UA -> Access OSBC -> SIPX

In the SipX we created a thrid Gateway (SIP TRUNK) set a sip prefix of 999,
point the the Peering ipaddr and the ITSP's SBC.
in the B2Bua we set sip:999* sip:x.x.x.241;ip-trunk=true ==> ITSP.

Our ITSP required only that the DID@Peering-ipaddr to gain access *****

and in the sip-trunk we set

route-set="ITSP-ipaddr"

sip-domain="Peering OSBC-ipaddr"

This is set to the Peering. Should this be set to our sipXecs Box?

*****

expires="10">

auth-user-name=""

auth-password=""
inbound-route="sip:2125551212@Peering OSBC-ipaddr"
expires="3600" />

Warren Kreckler

Original Message

From: "ehernaez"
To:
Sent: Saturday, December 29, 2007 9:58 AM
Subject: Re: OpenSIPStack OpenSBC Two (2) OSBC on net segment

Hi Warren,

Here is the call flow that you can use to bridge the 2 OSBC instances:
For an inbound call from PSTN: ITSP -> Peering OSBC -> Access

OSBC

-> SIPXFor an outbound cal to the PSTN: UA -> Access OSBC ->

SIPX -> Peering

OSBC -> ITSP
*inbound calls must be routed through the access OSBC in order to reach
endpoints that are registered thru the access OSBC
HTH

Guest

42. Jan 31, 2008 2:10 PM

in response to: Guest

Re: [OpenSIPStack] [OpenSBC] Two (2) OSBC on net segment

Hi ehernaez

Three questions from when i left for vacation; inbound (PSTN -> Peering
OSBC -> Access OSBC -> SIPX

1. In the Peering OSBC servers General settings, which Mode will best set
Peering? Since the Access must be set for B2Bua UpperRegister..

2. Linking the Peering OSBC to the Access OSBC, ideally what routes and
trust domains should be set on the Peering and Access OSBC?

3. Which of the two should have its SIP Trunk set. Assuming that inbouund
from PSTN will never need trunking to the Access OSBC?

The SIPX wiil have set two SIP Trunks one for inbound from and one for
outbound to PSTN/ITSP

Warren Kreckler

Original Message

From: "ehernaez"
To:
Sent: Saturday, December 29, 2007 9:58 AM
Subject: Re: OpenSIPStack OpenSBC Two (2) OSBC on net segment

Hi Warren,

Here is the call flow that you can use to bridge the 2 OSBC instances:
For an inbound call from PSTN: ITSP -> Peering OSBC -> Access

OSBC

-> SIPXFor an outbound cal to the PSTN: UA -> Access OSBC ->

SIPX -> Peering

OSBC -> ITSP
*inbound calls must be routed through the access OSBC in order to reach
endpoints that are registered thru the access OSBC
HTH

Guest

43. Mar 25, 2008 7:13 AM

in response to: Guest

Re: [OpenSIPStack] [ATLSIP] Ringing Sounds

This method does not work, because after it the method 1 is caused and
the sound does not interrupt. It can be necessary to add a variable 123
in class CallSession and to appropriate to it value TRUE in a method 1,
and in a method 2 to read out it and to appropriate to it value FALSE

This "quick hack" method does not work, because after

void SoftPhoneSIPEndPoint::OnProgress(...)
{
...
m_Manager.GetSoftPhoneInterface()->StopRingBackTone();
...
}

is called

void SoftPhoneSIPEndPoint::OnAlert(...)
{
...
m_Manager.GetSoftPhoneInterface()->PlayRingBackTone();
...
}

and the RingBackTone does not interrupt

It can be necessary (for example) to add a boolean variable m_NoRingBack
in class CallSession and to use it so:

void SoftPhoneSIPEndPoint::OnProgress(
CallSession & session,
const SIPMessage & alerting
)
{
...
if( session.GetType() == CallSession::Client )
session->m_NoRingBack = true;
...
}

void SoftPhoneSIPEndPoint::OnAlert(
CallSession & session,
const SIPMessage & alerting
)
{
...
if(!session->m_NoRingBack)
m_Manager.GetSoftPhoneInterface()->PlayRingBackTone();

session->m_NoRingBack = false;
...
}

Joegen E. Baclor wrote:

This may happen if your gateway sends a 180 without SDP followed by 180
or a 183 with SDP. This can be corrected by stopping the false ring
in OnProgress()

void SoftPhoneSIPEndPoint::OnProgress(
CallSession & session,
const SIPMessage & alerting
)
{
OpalOSSEndPoint::OnProgress( session, alerting );
PString info = session.GetTargetURI().AsString();
if( session.GetType() == CallSession::Client )
m_Manager.GetSoftPhoneInterface()->Event_OutgoingCallRinging( (const
char *)info );
}

Just insert m_Manager.GetSoftPhoneInterface()->StopRingBackTone(); in
this method as a quick hack. I think a cleaner way of doing this is to
not honor early media at all and retain the false ring if the call has
already received a no-media Alerting packet prior to the 183. perhaps
we can set this in the stack level. I am open to suggestions.

What's your view Ilian?

Joegen

Whit Thiele wrote:

Hey guys,

Where and when should the ringing sounds be generated? I use Asterisk so
when a call is launched, asterisk generates the ringing sound. If I don't
disable the PlayRingingSound methods, I get "double" rings.

Should this be a configurable setting in the ATLSIP library?

Whit

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Guest

44. Mar 25, 2008 7:15 AM

in response to: Guest

Re: [OpenSIPStack] [ATLSIP] Ringing Sounds

Sorry

This "quick hack" method does not work, because after

void SoftPhoneSIPEndPoint::OnProgress(...)
{
...
m_Manager.GetSoftPhoneInterface()->StopRingBackTone();
...
}

is called

void SoftPhoneSIPEndPoint::OnAlert(...)
{
...
m_Manager.GetSoftPhoneInterface()->PlayRingBackTone();
...
}

and the RingBackTone does not interrupt

Joegen E. Baclor wrote:

Should this be a configurable setting in the ATLSIP library?

Whit

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