asterisk users - Sep 2011 - Inter-astersik dialling encounteres no audio

I have been deploying Asterisk (open source PABX) in the company which I
work.

So far, all the Asterisk servers do not really talk to each other.
Recently, I am experimenting to dial from one Asterisk server to another
through the WAN and I encountered a no-audio problem although the
callee's phone can ring.
I understand that the no-audio means that SIP traffic (TCP/UDP 5060) is
allowed to go through but not RTP (UDP 16384-32767).
 
Case A
=====This is a simplified diagram of how I am testing the dialling between 2
subnets.
In this case, phone A is registered in Asterisk A and phone B is
registered in Asterisk B.

Phone A <--> Asterisk A <--> Router A <<==>> WAN
<<==>> Router B <-->
Asterisk B <--> Phone B   
 
Case B
=====However, before I have tested successfully using this kind of
connection.
In this case, phone B1 and B2 are registered in Asterisk B although they
are on different subnets.
Both phone B1 and B2 can ring and audio is allowed to pass through.

Phone B1 <--> Router A <<==>> WAN <<==>> Router B
<--> Asterisk B <-->
Phone B2
 
I am mystified why audio is allowed go through in case B but not case A.
 
Can someone be kind enough to help me to understand why I have this
problem?
If the router is blocking RTP traffic, then why is that I have no audio
problem in case B?

Thanks in advance.


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This obviously is pointing to NAT issue. see if you've configured both
asterisk servers with externip= PUBLICIPOFAsterisks.

Studying SIP traces on each console and specially looking at the SDPs in
INVITE will help you find out exact problem. I expect that one of the
asterisk box is sending the audio to its LAN/Private IP whereas it should be
sending RTPs to Public IP of other Asterisk.


On Fri, Sep 16, 2011 at 12:50 PM, Lee, John (Sydney) <John.Lee at
compuware.com
> wrote:
> **
>
> I have been deploying Asterisk (open source PABX) in the company which I
> work.
>
> So far, all the Asterisk servers do not really talk to each other.
> Recently, I am experimenting to dial from one Asterisk server to another
> through the WAN and I encountered a no-audio problem although the
callee's
> phone can ring.
>
> I understand that the no-audio means that SIP traffic (TCP/UDP 5060) is
> allowed to go through but not RTP (UDP 16384-32767).
>
>
>
> Case A
>
> =====>
> This is a simplified diagram of how I am testing the dialling between 2
> subnets.
>
> In this case, phone A is registered in Asterisk A and phone B is
> registered in Asterisk B.
>
> Phone A <--> Asterisk A <--> Router A <<==>> WAN
<<==>> Router B <-->
> Asterisk B <--> Phone B
>
>
>
> Case B
>
> =====>
> However, before I have tested successfully using this kind of connection.
>
> In this case, phone B1 and B2 are registered in Asterisk B although they
> are on different subnets.
>
> Both phone B1 and B2 can ring and audio is allowed to pass through.
>
> Phone B1 <--> Router A <<==>> WAN <<==>>
Router B <--> Asterisk B <-->
> Phone B2
>
>
>
> I am mystified why audio is allowed go through in case B but not case A.
>
>
>
> Can someone be kind enough to help me to understand why I have this
> problem?
>
> If the router is blocking RTP traffic, then why is that I have no audio
> problem in case B?
>
> Thanks in advance.
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
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>
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Lee, John (Sydney) wrote:
>
> I have been deploying Asterisk (open source PABX) in the company which I
work.
>
> Sofar, all the Asterisk servers do not really talk to each other. 
Recently, I am experimenting to dial from one Asterisk server to another through
the WAN and I encountered a no-audio problem although the callee's phone can
ring.
>
> I understand that the no-audio means that SIP traffic (TCP/UDP 5060) is
allowed to go through but not RTP (UDP 16384-32767).
>
> Case A
>
> =====>
> This is a simplified diagram of how I am testing the dialling between 2
subnets.
>
> In this case, phone A is registered in Asterisk A and phoneBis registered
in Asterisk B.
>
> Phone A <--> Asterisk A <--> Router A <<==>> WAN
<<==>> Router B <--> Asterisk B <--> Phone B
>
> Case B
>
> =====>
> However, before I have tested successfully using this kind of connection.
>
> In this case, phone B1 and B2 are registered in Asterisk B although they
are on different subnets.
>
> Both phone B1 and B2 can ring and audio is allowed to pass through.
>
> Phone B1 <--> Router A <<==>> WAN <<==>>
Router B <--> Asterisk B <--> Phone B2
>
> I am mystified why audio is allowed go through in case B but not case A.
>
> Can someone be kind enough to help me to understand why I have this
problem?
>
> If the router is blocking RTP traffic, then why is that I have no audio
problem in case B?
>
> Thanks in advance.
>
>
Why not use IAX????

John Novack

-- 

Dog is my Co-pilot

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Asterisk will send the two SIP endpoints 'reinvite' messages, so that
they talk RTP directly with each other.  Depending on your version of Asterisk,
setting the 'canreinvite' or 'directmedia' option may make a
difference, since that will keep the traffic flowing through the servers, and
the phones will not need to reach each other directly.






From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Lee, John (Sydney)
Sent: Friday, September 16, 2011 3:51 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Inter-astersik dialling encounteres no audio


I have been deploying Asterisk (open source PABX) in the company which I work.

So far, all the Asterisk servers do not really talk to each other.  Recently, I
am experimenting to dial from one Asterisk server to another through the WAN and
I encountered a no-audio problem although the callee's phone can ring.

I understand that the no-audio means that SIP traffic (TCP/UDP 5060) is allowed
to go through but not RTP (UDP 16384-32767).



Case A

=====
This is a simplified diagram of how I am testing the dialling between 2 subnets.

In this case, phone A is registered in Asterisk A and phone B is registered in
Asterisk B.

Phone A <--> Asterisk A <--> Router A <<==>> WAN
<<==>> Router B <--> Asterisk B <--> Phone B



Case B

=====
However, before I have tested successfully using this kind of connection.

In this case, phone B1 and B2 are registered in Asterisk B although they are on
different subnets.

Both phone B1 and B2 can ring and audio is allowed to pass through.

Phone B1 <--> Router A <<==>> WAN <<==>> Router B
<--> Asterisk B <--> Phone B2



I am mystified why audio is allowed go through in case B but not case A.



Can someone be kind enough to help me to understand why I have this problem?

If the router is blocking RTP traffic, then why is that I have no audio problem
in case B?

Thanks in advance.
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Thanks Sam, John and Justin for your wonderful advice.
Yes, it was the sip.conf parameter "reinvite=" which was causing the
problem.
Setting it to NO will fix it.	

Thanks all in asterisk-users mailing list.


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