Totally Stunned about this No Audio Going Out

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Totally Stunned about this No Audio Going Out

symack
Hello Everyone,

I had one way audo (out) for weeks, which was ok for our testing.
Plans were to integrate RTPProxy to manage two way audo, but
for now one way was ok.

Not sure what the hell I changed, and now I have no audio at all.
This is for an OpenSIPS -> Asterisk integration. I'm really streesed
about this and not sure where to start debugging this thing anymore.

Please Help,

Nick.

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Re: Totally Stunned about this No Audio Going Out

SamyGo
Hi,
I can imagine the gravity of the task you're stuck at. Can you take a sip trace, open it up in wireshark and see the SDPs which are exchanged between the two ends from OpenSIPS server. I'm pretty sure that if you didn't change anything then it must be the network/NAT changes which    resulted in no-audio at all. 
All I can imagine is that the two end points.. i.e Asterisk and Phone both trying to send Audio to Private subnet of other side thereby resulting in no-audio at all.

If your one end point had a public IP negotiated in SDP then you can expect at least one-way audio.

Take a SIP trace and reply here, maybe some other expert could comment then too.

Regards,
Sammy.

On Sat, Dec 17, 2011 at 2:43 AM, Nick Khamis <[hidden email]> wrote:
Hello Everyone,

I had one way audo (out) for weeks, which was ok for our testing.
Plans were to integrate RTPProxy to manage two way audo, but
for now one way was ok.

Not sure what the hell I changed, and now I have no audio at all.
This is for an OpenSIPS -> Asterisk integration. I'm really streesed
about this and not sure where to start debugging this thing anymore.

Please Help,

Nick.

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Re: Totally Stunned about this No Audio Going Out

symack
Hello Sammy,

Thank you for your response. I now have outgoing audio again which is
half the battle.
The second half (incoming audio), has proven to be a challenge. Maybe
if I start with
a description of the setup:

* This is a test environment done on virtual machines


Network:

RouterL (192.168.2.1)
Polycom Phone (192.168.2.11)
OpenSIPS (192.168.2.102)
Asterisk Virtual IP for AST1 and AST2 (192.168.2.6)
Asterisk1 (192.168.2.110)
Asterisk2 (192.168.2.111)


-------------  Port FWD (1)    --------------------------------
      ---------------------------
| Router |----------------------> |OpenSIPS/RTPProxy|----------> |
Asterisk GTWY  | ----------- Internet/ITSP
-------------
---------------------------------
---------------------------


1) The port forwarding range is:
     SIP: 5060
     RTP: 10,000-50,000
     RTP Proxy:  7789


I just want to clear some things up. I had outgoing audio the whole
time without RTPProxy.
All the test UC (Polycom Phones) are within the same network. Do I
need to use RTPProxy
to get incomming audio working? As you can see in the diagram, I did
try using RTP Proxy
but never succeeded.

Doing a raw UDP trace from ports (10000-50000) I found this:
http://pastebin.com/yzgBZQ9S
There is a "Destination unreachable" at first attempt being returned
by opensips server,
and then it dissapears, the it comes back again. Not sure if this is
related to the no
outgoing audio, but I will need to resolve it nevertheless.

As for a SIP trace without RTP Proxy proxy running:
http://pastebin.com/PUXJ3wpK.
Wanted to turn your attention to:

* The network architecture consists of OpenSIPS sending requests to
the Asterisk virtual IP (192.168.2.6),
which is connected to the Asterisk physical machines (192.168.2.110,
192.168.2.111). The responding
asterisk box, in this particular eaxample, was 192.168.2.111. I hope
this would not be the problem?

* A summary of the SDP trace is as follows:

INVITE from UC:                       m=audio 10006 RTP/AVP 0 8 18 101
OK from Asterisk to OpenSIPS: m=audio 31576 RTP/AVP 8 0 101.
OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.

Is taht my problem right there? My system is unable to connect
the initial request from the UC on port 10006, to the followup response
of the ITSP on port 34030? There is no OK from the UC to OpenSIPS.

I've been struggling with this for a week now. Any help would be greatly
appreciated!

Kind Regards,

Nick.

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Re: Totally Stunned about this No Audio Going Out

symack
What happened to my nice diagram? Argh.... Sorry guys!

Router -> OpenSIPS -> Asterisk -> ITPS

On Mon, Dec 19, 2011 at 12:20 PM, Nick Khamis <[hidden email]> wrote:

> Hello Sammy,
>
> Thank you for your response. I now have outgoing audio again which is
> half the battle.
> The second half (incoming audio), has proven to be a challenge. Maybe
> if I start with
> a description of the setup:
>
> * This is a test environment done on virtual machines
>
>
> Network:
>
> RouterL (192.168.2.1)
> Polycom Phone (192.168.2.11)
> OpenSIPS (192.168.2.102)
> Asterisk Virtual IP for AST1 and AST2 (192.168.2.6)
> Asterisk1 (192.168.2.110)
> Asterisk2 (192.168.2.111)
>
>
> -------------  Port FWD (1)    --------------------------------
>      ---------------------------
> | Router |----------------------> |OpenSIPS/RTPProxy|----------> |
> Asterisk GTWY  | ----------- Internet/ITSP
> -------------
> ---------------------------------
> ---------------------------
>
>
> 1) The port forwarding range is:
>     SIP: 5060
>     RTP: 10,000-50,000
>     RTP Proxy:  7789
>
>
> I just want to clear some things up. I had outgoing audio the whole
> time without RTPProxy.
> All the test UC (Polycom Phones) are within the same network. Do I
> need to use RTPProxy
> to get incomming audio working? As you can see in the diagram, I did
> try using RTP Proxy
> but never succeeded.
>
> Doing a raw UDP trace from ports (10000-50000) I found this:
> http://pastebin.com/yzgBZQ9S
> There is a "Destination unreachable" at first attempt being returned
> by opensips server,
> and then it dissapears, the it comes back again. Not sure if this is
> related to the no
> outgoing audio, but I will need to resolve it nevertheless.
>
> As for a SIP trace without RTP Proxy proxy running:
> http://pastebin.com/PUXJ3wpK.
> Wanted to turn your attention to:
>
> * The network architecture consists of OpenSIPS sending requests to
> the Asterisk virtual IP (192.168.2.6),
> which is connected to the Asterisk physical machines (192.168.2.110,
> 192.168.2.111). The responding
> asterisk box, in this particular eaxample, was 192.168.2.111. I hope
> this would not be the problem?
>
> * A summary of the SDP trace is as follows:
>
> INVITE from UC:                       m=audio 10006 RTP/AVP 0 8 18 101
> OK from Asterisk to OpenSIPS: m=audio 31576 RTP/AVP 8 0 101.
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
> OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
> OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
>
> Is taht my problem right there? My system is unable to connect
> the initial request from the UC on port 10006, to the followup response
> of the ITSP on port 34030? There is no OK from the UC to OpenSIPS.
>
> I've been struggling with this for a week now. Any help would be greatly
> appreciated!
>
> Kind Regards,
>
> Nick.

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Re: Totally Stunned about this No Audio Going Out

SamyGo
I think the time when you were having full audio w/o any RTP proxy was due to the fact that you've everything on the same subnet.
In my opinion you'll need  RTPproxy eventually whenever you deploy it in real environments.

I cant tell the exact reason for destination unreachable, maybe the virtual IP has something to do with it.

To configure RTP proxy you need to do following.
1- Start RTPproxy in bridged mode i.e #/usr/sbin/rtpproxy -l <externalipofproxy>/<internalipofproxy> blah blah switches
2- Set module params in opensips.cfg file, and find out the point in main route where call is forwarded to Asterisk Server VIP, just before that  write the function "force_rtp_proxy("ei")"
3- hmmm..on BYE or CANCEL you need to unforce rtp proxy as well.

Thats all I could think about it so far.  

Regards,
Sammy

On Mon, Dec 19, 2011 at 10:36 PM, Nick Khamis <[hidden email]> wrote:
What happened to my nice diagram? Argh.... Sorry guys!

Router -> OpenSIPS -> Asterisk -> ITPS

On Mon, Dec 19, 2011 at 12:20 PM, Nick Khamis <[hidden email]> wrote:
> Hello Sammy,
>
> Thank you for your response. I now have outgoing audio again which is
> half the battle.
> The second half (incoming audio), has proven to be a challenge. Maybe
> if I start with
> a description of the setup:
>
> * This is a test environment done on virtual machines
>
>
> Network:
>
> RouterL (192.168.2.1)
> Polycom Phone (192.168.2.11)
> OpenSIPS (192.168.2.102)
> Asterisk Virtual IP for AST1 and AST2 (192.168.2.6)
> Asterisk1 (192.168.2.110)
> Asterisk2 (192.168.2.111)
>
>
> -------------  Port FWD (1)    --------------------------------
>      ---------------------------
> | Router |----------------------> |OpenSIPS/RTPProxy|----------> |
> Asterisk GTWY  | ----------- Internet/ITSP
> -------------
> ---------------------------------
> ---------------------------
>
>
> 1) The port forwarding range is:
>     SIP: 5060
>     RTP: 10,000-50,000
>     RTP Proxy:  7789
>
>
> I just want to clear some things up. I had outgoing audio the whole
> time without RTPProxy.
> All the test UC (Polycom Phones) are within the same network. Do I
> need to use RTPProxy
> to get incomming audio working? As you can see in the diagram, I did
> try using RTP Proxy
> but never succeeded.
>
> Doing a raw UDP trace from ports (10000-50000) I found this:
> http://pastebin.com/yzgBZQ9S
> There is a "Destination unreachable" at first attempt being returned
> by opensips server,
> and then it dissapears, the it comes back again. Not sure if this is
> related to the no
> outgoing audio, but I will need to resolve it nevertheless.
>
> As for a SIP trace without RTP Proxy proxy running:
> http://pastebin.com/PUXJ3wpK.
> Wanted to turn your attention to:
>
> * The network architecture consists of OpenSIPS sending requests to
> the Asterisk virtual IP (192.168.2.6),
> which is connected to the Asterisk physical machines (192.168.2.110,
> 192.168.2.111). The responding
> asterisk box, in this particular eaxample, was 192.168.2.111. I hope
> this would not be the problem?
>
> * A summary of the SDP trace is as follows:
>
> INVITE from UC:                       m=audio 10006 RTP/AVP 0 8 18 101
> OK from Asterisk to OpenSIPS: m=audio 31576 RTP/AVP 8 0 101.
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
> OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
> OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
>
> Is taht my problem right there? My system is unable to connect
> the initial request from the UC on port 10006, to the followup response
> of the ITSP on port 34030? There is no OK from the UC to OpenSIPS.
>
> I've been struggling with this for a week now. Any help would be greatly
> appreciated!
>
> Kind Regards,
>
> Nick.

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Users mailing list
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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