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10 Replies Last post: Jul 28, 2009 12:46 AM by joegen  
 
dablackgreek
30 posts since
Jun 24, 2009

Jul 16, 2009 4:49 PM

OSBC Local UA and Remote UA Audio Problem


Hi All, I thought I had my OSBC system setup properly, but i later learn as i tested remote ua I was wrong.

I am having the following problem.

OSBC is in Front of Asterisk with B2BUppreg.

Local UA (IP Phones in the same subnet as OSBC & Asterisk) Register and can make calls perfectly.

Remote UA (IP Phones behind NAT in a remote network) Register and make calls but NO Audio is heard on either side

I observed that Xlite when register as Remote UA OSBC shows the public IP in the Registration BINDING, but when I used the Grandstream GXP2000 OSBC shows the private IP in the Registration BINDING

So the Xlite UA works fine becuz of the Public IP.....but the GXP2000 does not becuz of the Private IP


 
dablackgreek
30 posts since
Jun 24, 2009
1. Jul 16, 2009 7:34 PM in response to: dablackgreek
Re: OSBC Local UA and Remote UA Audio Problem
Additional observations:

When the Remote UA Calls the Local UA Audio is FINE, Works perfectly

When Local UA Calls the Remote UA NO Audio and also if the Remote UA side Hangs up the Local Side still acts like the call is active which means it didnt receive the BYE

I also confirm that its not a Grandstream GXP200 issue, I get the problem with a Sipura 3000 and Cisco 7960G.

I dont know why Xlite WORKS


Ideas anyone

 
uwaqar
6 posts since
Jul 8, 2009
2. Jul 16, 2009 7:33 PM in response to: dablackgreek
Re: OSBC Local UA and Remote UA Audio Problem

try Proxy-All-Media in the RTP settings

also, my experience so far shows that it always helps to reboot the SBC after manking any configuration changes


--

Umer

umer dot waqar at telepackets dot com

 
dablackgreek
30 posts since
Jun 24, 2009
3. Jul 16, 2009 7:39 PM in response to: uwaqar
Re: OSBC Local UA and Remote UA Audio Problem

OpenSBC WAN IP: w.x.y.188
OpenSBC LAN IP: 192.168.1.150

Asterisk IP: 192.168.1.250

[c-main]
RTTS-Client-Address=192.168.1.150

[c-main]
SIP-Log-Level=5
PTRACE-Log-Level=1
Log-File-Prefix=opensbc
User-Agent-Name=SBC
SBC-Application-Mode=B2BUpperReg Mode
Interface-Address Array Size=0
Enable-Backdoor-Port=True
Enable-Trunk-Port=False
Enable-Calea-Port=False
Static-RTP-Media-Address=
RTP-Min-Port=30001
RTP-Max-Port=40000
Send-Responses-Using-New-Socket=True
Enable-Local-Refer=False
Max-Forwards=70
Encryption-Mode=XOR
Encryption-Key=GS
Transaction-Thread-Count=10
Session-Thread-Count=10
Alerting-Timeout=30000
Seize-Timeout=60000
SIP-Timer-B=Default
SIP-Timer-H=Default
Session-Keep-Alive=1800
Session-Max-Life-Span=10800
Max-Concurrent-Session=100
Max-Call-Rate-Per-Second=10
Interface-Address 1=sip:192.168.1.150:5060

[c-main]
All-Reg-As-Upper-Reg=True
Enable-Stateful-Reg=False
Rewrite-TO-Domain=True
Rewrite-FROM-Domain=True
Route-List Array Size=1
Route-List 1=--sip:*@sip.mydomain.net-- sip:192.168.1.250;domain=sip.mydomain.com

[c-main]
Internal-DNS-Map Array Size=0

[c-main]
Proxy-On-Private-Contact=True
Proxy-On-via-received-vs-signaling-address=True
Proxy-On-Private-Via=True
Proxy-On-Different-RPORT=True
Proxy-All-Media=True

[c-main]
Enable-Media-Server=False
Media-Server-Number=5000
Codec-List Array Size=0
No-RTP-Proxy-On-All-Transfers=False
Enable-Announcement-Service=False
4xx-Error-Map=prompts/basic/cant_complete.wav
5xx-Error-Map=prompts/basic/cant_complete.wav
6xx-Error-Map=prompts/basic/cant_complete.wav
Announcement-Error-Map Array Size=0

[c-main]
Enable-Route-Scripting=False
Route-Script=b2bua-route.cscript
Route-List Array Size=1
Route-List 1=--sip:*@sip.mydomain.net-- sip:192.168.1.250
Catch-All-Route=
Insert-Route-Header=True
Rewrite-TO-URI=True
Rewrite-From-Domain=
Prepend-ISUP-OLI=False
Route-By-Request-URI=True
Route-By-To-URI=False
Drop-Routes-On-Ping-Timeout=False
Use-External-XML=False
External-XML-File=b2bua-route.xml

[c-main]
Accept-All-Registration=True
Account-List Array Size=0

[c-main]
Accept-All-Calls=True
Trusted-Domain-List Array Size=0

[c-main]
Default-Realm=
Trusted-Domain-List Array Size=0

[c-main]
Main-Interface-Address Array Size=2
Backdoor-Interface-Address=sip:*:5062
Trunk-Interface-Address=sip:*:5064
Media-Server-Interface-Address=sip:*:5066
CALEA-Interface-Address=sip:*:5068
Auxiliary-Interface-Address=sip:*:5070
Interface-Route-List Array Size=2
Interface-Route-List 1=--sip:192.168.1.*-- sip:192.168.1.150
Main-Interface-Address 1=sip:192.168.1.150:5060
Main-Interface-Address 2=sip:w.x.y.188:5060
Interface-Route-List 2=--sip:w.x.y.*-- sip:192.168.1.150;external=w.x.y.188

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joegen
519 posts since
Apr 28, 2007
4. Jul 16, 2009 8:01 PM in response to: dablackgreek
Re: OSBC Local UA and Remote UA Audio Problem
{quote:title=dablackgreek wrote:}{quote}

When Local UA Calls the Remote UA NO Audio and also if the Remote UA side Hangs up the Local Side still acts like the call is active which means it didnt receive the BYE

I also confirm that its not a Grandstream GXP200 issue, I get the problem with a Sipura 3000 and Cisco 7960G.

I dont know why Xlite WORKS
The BYE issue is easy to figure out. just check the logs if OpenSBC receives the BYE and what it does with it. The audio issue is tough beast to crack. First item in your list is to bring out wireshark and shiff RTP and make sure that RTP flows in both directions. The next thing I would do is to look at the SDP and see if your UA is trying to perform near-end NAT traversal on its own. You can't mix far-end and near-end. You have to choose one approach and stick with it. If you have eliminated this possibility, then the next thing to do is to see if your UA is using the same port to send and receive RTP. OSBC media proxy would only work if send and receive port for RTP is the same. There are implementations that lets you choose whether to use the same port for receiving and sending or not.
 
dablackgreek
30 posts since
Jun 24, 2009
5. Jul 16, 2009 8:27 PM in response to: joegen
Re: OSBC Local UA and Remote UA Audio Problem
Joegen, is there an explanation for why XLite UA would register with its Public IP and the GXP2000, Sipura3000 and Ciso 7960 and registering with the Private IP..

The remote Network is the same that i am testing all the UA's with


I sent LOGS to your PM... Look at the RTP Statistics

 
dablackgreek
30 posts since
Jun 24, 2009
6. Jul 17, 2009 4:42 PM in response to: dablackgreek
Re: OSBC Local UA and Remote UA Audio Problem

Got my audio problem resolved... See changes i made to Sip Transports to ensure proper routing

OSBC is very funny with its routing...

[SIP-Transports]
Main-Interface-Address Array Size=2
Backdoor-Interface-Address=sip:192.168.1.150:5062
Trunk-Interface-Address=sip:*:5064
Media-Server-Interface-Address=sip:*:5066
CALEA-Interface-Address=sip:*:5068
Auxiliary-Interface-Address=sip:*:5070
Interface-Route-List Array Size=2
Main-Interface-Address 1=sip:192.168.1.150:5060
Main-Interface-Address 2=sip:w.x.y.188:5060
Interface-Route-List 1=[sip:*] sip:w.x.y.188
Interface-Route-List 2=[sip:192.168.1.*] sip:192.168.1.150;external=w.x.y.188

 
isaacx
4 posts since
Jul 22, 2009
7. Jul 23, 2009 3:46 AM in response to: dablackgreek
Re: OSBC Local UA and Remote UA Audio Problem

Hi [~723],

I have the same problem of audio problem. extension registered in the internal private network (same as Asterisk pbx) can't make call to extension behind NAT registered throught OSBC.

outside xlite software phone extension encountered one way audio, but OSSPHONE works fine.

I followed your instrustion to add route in SIP-Transports. but still make call to external extension. would anyone help? thanks.

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joegen
519 posts since
Apr 28, 2007
8. Jul 23, 2009 9:14 PM in response to: isaacx
Re: OSBC Local UA and Remote UA Audio Problem
{quote:title=isaacx wrote:}{quote}

Hi [~723],

I have the same problem of audio problem. extension registered in the internal private network (same as Asterisk pbx) can't make call to extension behind NAT registered throught OSBC.


Can't make call to extension is pretty vague. What exactly ahppens to the INVITE once it reaches OpenSBC? Responded with a 404? If so, then you need to correct something in routing.


outside xlite software phone extension encountered one way audio, but OSSPHONE works fine.

If OSSPhone works and XLite doesnt, then this is surely a sign of a misconfiguration on XLite. Make sure that XLite ICE support and all other near-end NAT traversal options are off.


I followed your instrustion to add route in SIP-Transports. but still make call to external extension. would anyone help? thanks.

 
isaacx
4 posts since
Jul 22, 2009
9. Jul 27, 2009 8:10 PM in response to: joegen
Re: OSBC Local UA and Remote UA Audio Problem
I found that it was my Asterisk PBX near end Netscreen firewall ALG problem. If I disable Netscreen ALG, it works fine. However I can't disable it as my Asterisk box has a SIP trunk to Voicetrading VoIP service provider (if ALG disabled, trunk to VoIP provider not working). :(. Anything I can do on OSBC side?
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joegen
519 posts since
Apr 28, 2007
10. Jul 28, 2009 12:46 AM in response to: isaacx
Re: OSBC Local UA and Remote UA Audio Problem

Attach separate level 5 logs of the calls with and without the ALG respectively. Let's check what we could find out from the SIP headers.

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