[mobicents-public] IBPX PSTN call
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Wagner Andrade
May 4, 2010, 2:16:26 PM5/4/10
to mobicents-public
Hi,
I'm testing the Media IPBX project (http://www.mobicents.org/mss-
ipbx.html) and all works fine.
But when I calling to a mobile phone, using PSTN configurations (with
my SIP provider), the phone rings, I attend, the connection is
established but I can't listen nothing in both sides (softphone and
mobile).
(local call with softphones works fine)
Someone that tested had this problem?
I'm testing the Media IPBX project (http://www.mobicents.org/mss-
ipbx.html) and all works fine.
But when I calling to a mobile phone, using PSTN configurations (with
my SIP provider), the phone rings, I attend, the connection is
established but I can't listen nothing in both sides (softphone and
mobile).
(local call with softphones works fine)
Someone that tested had this problem?
Vladimir Ralev
May 4, 2010, 2:32:29 PM5/4/10
to mobicent...@googlegroups.com
Most likely a NAT problem. Are you behind NAT or some firewall? There
are a couple of things you can do about it. IPBX, MMS and MSS can be
configured to work behind NAT (MMS patch is needed) or you can change
your PSTN provider with some other that works around the NAT problems.
are a couple of things you can do about it. IPBX, MMS and MSS can be
configured to work behind NAT (MMS patch is needed) or you can change
your PSTN provider with some other that works around the NAT problems.
Wagner Andrade
May 4, 2010, 3:56:24 PM5/4/10
to mobicents-public
Yes, I'm behind NAT and firewall.
I don't understand why my mobile phone rings, but I can't listen any
sound.
MMS patch that you said is stun server?
On 4 maio, 15:32, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> Most likely a NAT problem. Are you behind NAT or some firewall? There
> are a couple of things you can do about it. IPBX, MMS and MSS can be
> configured to work behind NAT (MMS patch is needed) or you can change
> your PSTN provider with some other that works around the NAT problems.
>
> On Tue, May 4, 2010 at 9:16 PM, Wagner Andrade
>
I don't understand why my mobile phone rings, but I can't listen any
sound.
MMS patch that you said is stun server?
On 4 maio, 15:32, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> Most likely a NAT problem. Are you behind NAT or some firewall? There
> are a couple of things you can do about it. IPBX, MMS and MSS can be
> configured to work behind NAT (MMS patch is needed) or you can change
> your PSTN provider with some other that works around the NAT problems.
>
> On Tue, May 4, 2010 at 9:16 PM, Wagner Andrade
>
Vladimir Ralev
May 4, 2010, 4:13:48 PM5/4/10
to mobicent...@googlegroups.com
It depends on your PSTN gateway, the patch might be needed or not.
Some PSTN providers are capable of sending RTP stream to the same
address/port where the receive from (this is also what the MMS patch
does). What PSTN gateway do you use? Most have settings to compensate
lack of traversal and you can have normal calls (callwithus). Your
phone rings because some SIP messages pass thru, but RTP may be not.
Unfortunately it is also a fact that different PSTN networks around
the world behave differently. For testing I suggest to use USA toll
free numbers such as google search. Then try to use your network and
if it doesnt work, see what is different.
This is the patch
http://code.google.com/p/mobicents/issues/detail?id=854 for MMS for
RTP
For SIP you can configure MSS STUN settings.
Some PSTN providers are capable of sending RTP stream to the same
address/port where the receive from (this is also what the MMS patch
does). What PSTN gateway do you use? Most have settings to compensate
lack of traversal and you can have normal calls (callwithus). Your
phone rings because some SIP messages pass thru, but RTP may be not.
Unfortunately it is also a fact that different PSTN networks around
the world behave differently. For testing I suggest to use USA toll
free numbers such as google search. Then try to use your network and
if it doesnt work, see what is different.
This is the patch
http://code.google.com/p/mobicents/issues/detail?id=854 for MMS for
RTP
For SIP you can configure MSS STUN settings.
Wagner Andrade
May 4, 2010, 4:34:09 PM5/4/10
to mobicents-public
Makes sense!
Using a Sniffer I can see the RTP packeges traveling in the local
call, but don't happens in a mobile call.
Thanks, I will try and give feedback.
On 4 maio, 17:13, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> It depends on your PSTN gateway, the patch might be needed or not.
> Some PSTN providers are capable of sending RTP stream to the same
> address/port where the receive from (this is also what the MMS patch
> does). What PSTN gateway do you use? Most have settings to compensate
> lack of traversal and you can have normal calls (callwithus). Your
> phone rings because some SIP messages pass thru, but RTP may be not.
> Unfortunately it is also a fact that different PSTN networks around
> the world behave differently. For testing I suggest to use USA toll
> free numbers such as google search. Then try to use your network and
> if it doesnt work, see what is different.
>
> This is the patchhttp://code.google.com/p/mobicents/issues/detail?id=854for MMS for
Using a Sniffer I can see the RTP packeges traveling in the local
call, but don't happens in a mobile call.
Thanks, I will try and give feedback.
On 4 maio, 17:13, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> It depends on your PSTN gateway, the patch might be needed or not.
> Some PSTN providers are capable of sending RTP stream to the same
> address/port where the receive from (this is also what the MMS patch
> does). What PSTN gateway do you use? Most have settings to compensate
> lack of traversal and you can have normal calls (callwithus). Your
> phone rings because some SIP messages pass thru, but RTP may be not.
> Unfortunately it is also a fact that different PSTN networks around
> the world behave differently. For testing I suggest to use USA toll
> free numbers such as google search. Then try to use your network and
> if it doesnt work, see what is different.
>
> RTP
> For SIP you can configure MSS STUN settings.
>
> On Tue, May 4, 2010 at 10:56 PM, Wagner Andrade
>
> For SIP you can configure MSS STUN settings.
>
> On Tue, May 4, 2010 at 10:56 PM, Wagner Andrade
>
Wagner Andrade
May 5, 2010, 12:30:27 PM5/5/10
to mobicents-public
Hi Vladimir,
How can I apply this patch in Mobicents stable version (JBoss 4.2)?
I tried run IPBX project in the last snapshot, withJBoss 5.0, but it's
fail.
How can I apply this patch in Mobicents stable version (JBoss 4.2)?
I tried run IPBX project in the last snapshot, withJBoss 5.0, but it's
fail.
> It depends on your PSTN gateway, the patch might be needed or not.
> Some PSTN providers are capable of sending RTP stream to the same
> address/port where the receive from (this is also what the MMS patch
> does). What PSTN gateway do you use? Most have settings to compensate
> lack of traversal and you can have normal calls (callwithus). Your
> phone rings because some SIP messages pass thru, but RTP may be not.
> Unfortunately it is also a fact that different PSTN networks around
> the world behave differently. For testing I suggest to use USA toll
> free numbers such as google search. Then try to use your network and
> if it doesnt work, see what is different.
>
> This is the patchhttp://code.google.com/p/mobicents/issues/detail?id=854for MMS for
> Some PSTN providers are capable of sending RTP stream to the same
> address/port where the receive from (this is also what the MMS patch
> does). What PSTN gateway do you use? Most have settings to compensate
> lack of traversal and you can have normal calls (callwithus). Your
> phone rings because some SIP messages pass thru, but RTP may be not.
> Unfortunately it is also a fact that different PSTN networks around
> the world behave differently. For testing I suggest to use USA toll
> free numbers such as google search. Then try to use your network and
> if it doesnt work, see what is different.
>
> RTP
> For SIP you can configure MSS STUN settings.
>
> On Tue, May 4, 2010 at 10:56 PM, Wagner Andrade
>
> For SIP you can configure MSS STUN settings.
>
> On Tue, May 4, 2010 at 10:56 PM, Wagner Andrade
>
Wagner Andrade
May 10, 2010, 1:34:35 PM5/10/10
to mobicents-public
That issue happens just behind NAT?
(http://code.google.com/p/mobicents/issues/detail?id=854)
I installed mobicents in a host that don't use NAT, nevertheless I
can't listen any audio in a PSTN call.
On 5 maio, 13:30, Wagner Andrade <wagner.andradesi...@gmail.com>
wrote:
(http://code.google.com/p/mobicents/issues/detail?id=854)
I installed mobicents in a host that don't use NAT, nevertheless I
can't listen any audio in a PSTN call.
On 5 maio, 13:30, Wagner Andrade <wagner.andradesi...@gmail.com>
wrote:
> Hi Vladimir,
>
> How can I apply this patch in Mobicents stable version (JBoss 4.2)?
> I tried run IPBX project in the last snapshot, withJBoss 5.0, but it's
> fail.
>
> On 4 maio, 17:13, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
>
> > It depends on your PSTN gateway, the patch might be needed or not.
> > Some PSTN providers are capable of sending RTP stream to the same
> > address/port where the receive from (this is also what the MMS patch
> > does). What PSTN gateway do you use? Most have settings to compensate
> > lack of traversal and you can have normal calls (callwithus). Your
> > phone rings because some SIP messages pass thru, but RTP may be not.
> > Unfortunately it is also a fact that different PSTN networks around
> > the world behave differently. For testing I suggest to use USA toll
> > free numbers such as google search. Then try to use your network and
> > if it doesnt work, see what is different.
>
> > This is the patchhttp://code.google.com/p/mobicents/issues/detail?id=854forMMS for
>
> How can I apply this patch in Mobicents stable version (JBoss 4.2)?
> I tried run IPBX project in the last snapshot, withJBoss 5.0, but it's
> fail.
>
> On 4 maio, 17:13, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
>
> > It depends on your PSTN gateway, the patch might be needed or not.
> > Some PSTN providers are capable of sending RTP stream to the same
> > address/port where the receive from (this is also what the MMS patch
> > does). What PSTN gateway do you use? Most have settings to compensate
> > lack of traversal and you can have normal calls (callwithus). Your
> > phone rings because some SIP messages pass thru, but RTP may be not.
> > Unfortunately it is also a fact that different PSTN networks around
> > the world behave differently. For testing I suggest to use USA toll
> > free numbers such as google search. Then try to use your network and
> > if it doesnt work, see what is different.
>
Vladimir Ralev
May 10, 2010, 1:40:53 PM5/10/10
to mobicent...@googlegroups.com
Have you identified where the voice is lost? Can we see the wireshark trace?
Wagner Andrade
May 10, 2010, 2:15:31 PM5/10/10
to mobicents-public
Now mobicents is installed in a remote host, I can't use wareshark
there.
But the behavior is exacly the same when installed in my local
computer (that uses NAT).
I use wareshark in my PC and realized that no exists traffic of RTP
package during the PSTN call.
On 10 maio, 14:40, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> Have you identified where the voice is lost? Can we see the wireshark trace?
>
> On Mon, May 10, 2010 at 8:34 PM, Wagner Andrade <
>
there.
But the behavior is exacly the same when installed in my local
computer (that uses NAT).
I use wareshark in my PC and realized that no exists traffic of RTP
package during the PSTN call.
On 10 maio, 14:40, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> Have you identified where the voice is lost? Can we see the wireshark trace?
>
> On Mon, May 10, 2010 at 8:34 PM, Wagner Andrade <
>
Vladimir Ralev
May 10, 2010, 2:19:20 PM5/10/10
to mobicent...@googlegroups.com
OK, can you send us your local wireshark then? Also did you try to use google phone number for test, do all PSTN networks behave the same way?
Wagner Andrade
May 10, 2010, 3:58:49 PM5/10/10
to mobicents-public
Firstly thanks for your help, Vladimir.
I'll be more directly on my need.
I'm looking a SIP plataform that I will use in my college work, this
work need a integration with PSTN network.
My first choices are Mobicents and Sailfin, and in a fast analysis,
I'm giving preference for Mobicents for the maturity of project.
The first test for realy elect the plataform is the most important
feature: integration with PSTN.
In my tests I can see that this integration of Mobicents works (making
a call to my mobile phone), but I can't hear any audio.
For do this I use the IPBX example because is a complete project, so
is more fast to get a concrete result.
Now my question is diferent.
There is another way (easy and fast) to test the integration of
Mobicents with PSTN network?
I just need see it working to continue my work in the correct
plataform.
Thanks so much for the attention.
On 10 maio, 15:19, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> OK, can you send us your local wireshark then? Also did you try to use
> google phone number for test, do all PSTN networks behave the same way?
>
> On Mon, May 10, 2010 at 9:15 PM, Wagner Andrade <
>
I'll be more directly on my need.
I'm looking a SIP plataform that I will use in my college work, this
work need a integration with PSTN network.
My first choices are Mobicents and Sailfin, and in a fast analysis,
I'm giving preference for Mobicents for the maturity of project.
The first test for realy elect the plataform is the most important
feature: integration with PSTN.
In my tests I can see that this integration of Mobicents works (making
a call to my mobile phone), but I can't hear any audio.
For do this I use the IPBX example because is a complete project, so
is more fast to get a concrete result.
Now my question is diferent.
There is another way (easy and fast) to test the integration of
Mobicents with PSTN network?
I just need see it working to continue my work in the correct
plataform.
Thanks so much for the attention.
On 10 maio, 15:19, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> OK, can you send us your local wireshark then? Also did you try to use
> google phone number for test, do all PSTN networks behave the same way?
>
> On Mon, May 10, 2010 at 9:15 PM, Wagner Andrade <
>
Wagner Andrade
May 10, 2010, 3:58:56 PM5/10/10
to mobicents-public
Firstly thanks for your help, Vladimir.
I'll be more directly on my need.
I'm looking a SIP plataform that I will use in my college work, this
work need a integration with PSTN network.
My first choices are Mobicents and Sailfin, and in a fast analysis,
I'm giving preference for Mobicents for the maturity of project.
The first test for realy elect the plataform is the most important
feature: integration with PSTN.
In my tests I can see that this integration of Mobicents works (making
a call to my mobile phone), but I can't hear any audio.
For do this I use the IPBX example because is a complete project, so
is more fast to get a concrete result.
Now my question is diferent.
There is another way (easy and fast) to test the integration of
Mobicents with PSTN network?
I just need see it working to continue my work in the correct
plataform.
Thanks so much for the attention.
On 10 maio, 15:19, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
I'll be more directly on my need.
I'm looking a SIP plataform that I will use in my college work, this
work need a integration with PSTN network.
My first choices are Mobicents and Sailfin, and in a fast analysis,
I'm giving preference for Mobicents for the maturity of project.
The first test for realy elect the plataform is the most important
feature: integration with PSTN.
In my tests I can see that this integration of Mobicents works (making
a call to my mobile phone), but I can't hear any audio.
For do this I use the IPBX example because is a complete project, so
is more fast to get a concrete result.
Now my question is diferent.
There is another way (easy and fast) to test the integration of
Mobicents with PSTN network?
I just need see it working to continue my work in the correct
plataform.
Thanks so much for the attention.
On 10 maio, 15:19, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> OK, can you send us your local wireshark then? Also did you try to use
> google phone number for test, do all PSTN networks behave the same way?
>
> On Mon, May 10, 2010 at 9:15 PM, Wagner Andrade <
>
> google phone number for test, do all PSTN networks behave the same way?
>
> On Mon, May 10, 2010 at 9:15 PM, Wagner Andrade <
>
Vladimir Ralev
May 10, 2010, 4:36:53 PM5/10/10
to mobicent...@googlegroups.com
There are a couple of ways to talk to PSTN network. PSTN gateways are the easiest, but with Mobicents you could also do SS7.
You should note that IPBX architecture is slightly more complicated than just a PSTN call - it uses conferences and terminates the calls instead of forwarding them.
Forwarding the call is much easier and it will work with any PSTN gateway once you adjust it on the SIP level.
Terminating the call like IPBX does requires adjustments at both SIP and RTP levels, which is a bit more complicated, but anything can be done.
We have tested IPBX with 2 PSTN providers behind a firewall and they work with most PSTN network, but some PSTN networks are known to have problems.
The easiest way to test it is to take IPBX, there is a preconfigured callwithus account in there and you can make a test call to google phone, which is already in the contact list of the default users. Dont make more than a few calls to a cheap network. This account is intended only for a quick test.
A wireshark trace will tell us a lot more about your situation. I can't see any reason MMS would not start sending RTP.
Wagner Andrade
May 11, 2010, 9:45:27 AM5/11/10
to mobicents-public
Thanks for you information about forward call, it's a feature that my
work will implement.
When you said that some PSTN networks had problems, means that I can't
use Mobicents to comunicates with this networks or that need a more
complex control of my application?
I will create the wireshark trace again and send to you today.
On 10 maio, 17:36, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> There are a couple of ways to talk to PSTN network. PSTN gateways are the
> easiest, but with Mobicents you could also do SS7.
>
> You should note that IPBX architecture is slightly more complicated than
> just a PSTN call - it uses conferences and terminates the calls instead of
> forwarding them.
>
> Forwarding the call is much easier and it will work with any PSTN gateway
> once you adjust it on the SIP level.
> Terminating the call like IPBX does requires adjustments at both SIP and RTP
> levels, which is a bit more complicated, but anything can be done.
>
> We have tested IPBX with 2 PSTN providers behind a firewall and they work
> with most PSTN network, but some PSTN networks are known to have problems.
>
> The easiest way to test it is to take IPBX, there is a preconfigured
> callwithus account in there and you can make a test call to google phone,
> which is already in the contact list of the default users. Dont make more
> than a few calls to a cheap network. This account is intended only for a
> quick test.
>
> A wireshark trace will tell us a lot more about your situation. I can't see
> any reason MMS would not start sending RTP.
>
> On Mon, May 10, 2010 at 10:58 PM, Wagner Andrade <
>
work will implement.
When you said that some PSTN networks had problems, means that I can't
use Mobicents to comunicates with this networks or that need a more
complex control of my application?
I will create the wireshark trace again and send to you today.
On 10 maio, 17:36, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> There are a couple of ways to talk to PSTN network. PSTN gateways are the
> easiest, but with Mobicents you could also do SS7.
>
> You should note that IPBX architecture is slightly more complicated than
> just a PSTN call - it uses conferences and terminates the calls instead of
> forwarding them.
>
> Forwarding the call is much easier and it will work with any PSTN gateway
> once you adjust it on the SIP level.
> Terminating the call like IPBX does requires adjustments at both SIP and RTP
> levels, which is a bit more complicated, but anything can be done.
>
> We have tested IPBX with 2 PSTN providers behind a firewall and they work
> with most PSTN network, but some PSTN networks are known to have problems.
>
> The easiest way to test it is to take IPBX, there is a preconfigured
> callwithus account in there and you can make a test call to google phone,
> which is already in the contact list of the default users. Dont make more
> than a few calls to a cheap network. This account is intended only for a
> quick test.
>
> A wireshark trace will tell us a lot more about your situation. I can't see
> any reason MMS would not start sending RTP.
>
> On Mon, May 10, 2010 at 10:58 PM, Wagner Andrade <
>
Wagner Andrade
May 11, 2010, 5:02:58 PM5/11/10
to mobicents-public
I used the Alerting Application example (http://www.mobicents.org/
alerting-app.html) to simulate a diferent cenario for test, did few
modifications for work with my PSTN gateway.
Again, my mobile phone rings, I received de call but didn't hear any
audio.
I did get the wireshark trace of this test, but has private
informations about my gateway.
Can I send this trace to your email?
alerting-app.html) to simulate a diferent cenario for test, did few
modifications for work with my PSTN gateway.
Again, my mobile phone rings, I received de call but didn't hear any
audio.
I did get the wireshark trace of this test, but has private
informations about my gateway.
Can I send this trace to your email?
> There are a couple of ways to talk to PSTN network. PSTN gateways are the
> easiest, but with Mobicents you could also do SS7.
>
> You should note that IPBX architecture is slightly more complicated than
> just a PSTN call - it uses conferences and terminates the calls instead of
> forwarding them.
>
> Forwarding the call is much easier and it will work with any PSTN gateway
> once you adjust it on the SIP level.
> Terminating the call like IPBX does requires adjustments at both SIP and RTP
> levels, which is a bit more complicated, but anything can be done.
>
> We have tested IPBX with 2 PSTN providers behind a firewall and they work
> with most PSTN network, but some PSTN networks are known to have problems.
>
> The easiest way to test it is to take IPBX, there is a preconfigured
> callwithus account in there and you can make a test call to google phone,
> which is already in the contact list of the default users. Dont make more
> than a few calls to a cheap network. This account is intended only for a
> quick test.
>
> A wireshark trace will tell us a lot more about your situation. I can't see
> any reason MMS would not start sending RTP.
>
> On Mon, May 10, 2010 at 10:58 PM, Wagner Andrade <
>
> easiest, but with Mobicents you could also do SS7.
>
> You should note that IPBX architecture is slightly more complicated than
> just a PSTN call - it uses conferences and terminates the calls instead of
> forwarding them.
>
> Forwarding the call is much easier and it will work with any PSTN gateway
> once you adjust it on the SIP level.
> Terminating the call like IPBX does requires adjustments at both SIP and RTP
> levels, which is a bit more complicated, but anything can be done.
>
> We have tested IPBX with 2 PSTN providers behind a firewall and they work
> with most PSTN network, but some PSTN networks are known to have problems.
>
> The easiest way to test it is to take IPBX, there is a preconfigured
> callwithus account in there and you can make a test call to google phone,
> which is already in the contact list of the default users. Dont make more
> than a few calls to a cheap network. This account is intended only for a
> quick test.
>
> A wireshark trace will tell us a lot more about your situation. I can't see
> any reason MMS would not start sending RTP.
>
> On Mon, May 10, 2010 at 10:58 PM, Wagner Andrade <
>
Vladimir Ralev
May 11, 2010, 6:21:55 PM5/11/10
to mobicent...@googlegroups.com
OK please send it to me by email.
Again, this is the same issue. RTP adjustment is a challenge for any media server. Keep in mind that you can use Mobicents Sip Servlets with any other media server for RTP if MMS doesn't work for you. SIP and RTP are different topics, different servers (MSS and MMS). Support is per standard and responsisbility for standards lies in both Mobicents and the PSTN gateway providers. Some networks are known to hae problems because certain telco operators filter VoIP traffic or ue gateways that are not standard compliant. Sometimes it behaves randomly because of round-robin plicies at the PSTN gateway side. And many many other problems are possible.
Wagner Andrade
May 12, 2010, 8:51:35 AM5/12/10
to mobicents-public
Thanks for your help, Vladimir.
I'm send wireshark trace.
My IP is [192.168.0.163] and [201.86.87.5] is the gateway
(vono.net.br).
The package number 4 is the begining of call trace.
BYE request is sended of my mobile phone, in package 66.
Call duration was about 15 secounds.
Realize that don't has any RTP package.
Send to me any question.
On 11 maio, 19:21, Vladimir Ralev <vladimir.ra...@gmail.com> wrote:
> OK please send it to me by email.
>
> Again, this is the same issue. RTP adjustment is a challenge for any media
> server. Keep in mind that you can use Mobicents Sip Servlets with any other
> media server for RTP if MMS doesn't work for you. SIP and RTP are different
> topics, different servers (MSS and MMS). Support is per standard and
> responsisbility for standards lies in both Mobicents and the PSTN gateway
> providers. Some networks are known to hae problems because certain telco
> operators filter VoIP traffic or ue gateways that are not standard
> compliant. Sometimes it behaves randomly because of round-robin plicies at
> the PSTN gateway side. And many many other problems are possible.
>
> On Wed, May 12, 2010 at 12:02 AM, Wagner Andrade <
>
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