LanScape Support Forum: Need Help regarding Media Proxy and Centrex

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SIP Proxy and Media Proxy - Pre-Sales Technical Support
LanScape Support Forum -> SIP Proxy and Media Proxy - Pre-Sales Technical Support

Topic: Need Help regarding Media Proxy and Centrex

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khurram
Intermediate

Joined: October 06 2008
Location: Pakistan
Posts: 5

Posted: October 06 2008 at 8:14am \| IP Logged

i have download the Media Proxy and Centrex Proxy and configured them. they are communicating with each other..

and i am using third party(Celliance) SIP Client with video support to communicate with the centrex sip proxy.
it registers well withourt any error

i try to make a call to another sip client without sdp support it connects call fails and when i try to make a call to another sip client with sdp support it connects but neither i can hear nor cant see video.

i want to make a sip call with audio and video support..

Plz help me if any prob with the lanscape configuration or third party sip client issue

Regards

Khurram Shakoor

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: October 06 2008 at 12:46pm \| IP Logged

Hi khurram,

Thanks for posting and waiting for this response.

Whatever the issue, its probably something simple associated with the SIP (SIP inter-op issue).

We would love to work with you and get this audio+video SIP client working with our proxies.

Preferred Method:
If it is possible to get one of the Celliance SIP clients from you, we can perform the basic audio and video call testing to determine what is going on. We would do this at our location.

Non-preferred method:
The other possibility is for you to post SIP logs for the calls that are failing and we will have to pick through the SIP to see what is occurring. This will take longer and we would rather test using one of your live SIP clients.

Once we find out why audio and video are not working with this SIP client, you will really like how the LanScape proxies work. They are simple and perform extremely well.

Let us know what you want to do.

By the way, how large is your deployment (how many proxies would you be purchasing?).

Thanks again and repost as needed,

Support

khurram
Intermediate

Joined: October 06 2008
Location: Pakistan
Posts: 5

Posted: October 06 2008 at 10:55pm \| IP Logged

Thank you for your reply. i need two to three proxies for sip and media. i am attaching the sip sdk.

plz reply me soon because our project is already late.

plz tell me where should i upload/email the sip client sdk...

thank you.

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: October 07 2008 at 8:04am \| IP Logged

Hi khurram,

Sorry your project is late. We will do what we can.

We will send you an email that contains your support FTP account info. Upload your SIP client (or SDK) and post back here when you are done.

Support

khurram
Intermediate

Joined: October 06 2008
Location: Pakistan
Posts: 5

Posted: October 08 2008 at 12:33am \| IP Logged

Hi Admin

I have uploaded the SIP Client SDK at the FTP Server, you email to me.

Just give simple sip account settings to logon.

The sdk i am giving you is application of .NET(C#)

waiting for your reply..

Regards

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: October 08 2008 at 5:17am \| IP Logged

Hi khurram,

We will look at it today and repost with any questions if needed.

Hang on...

Support

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: October 08 2008 at 8:22am \| IP Logged

Hi khurram,

We performed basic testing using your PortSIP SDK and the SIPSample C# sample application.

From what we see, the LanScape Centrex SIP proxy + LanScape VOIP Media proxy are functioning normally.

We set up this test scenario:

SIPSample user agent #1:
Extension 102
Registers OK
Configured to use H264 video and uLaw audio.
IP address: 192.168.1.2
SIP port: 5560

SIPSample user agent #2:
Extension 103
Registers OK
Configured to use H264 video and uLaw audio.
IP address: 192.168.1.80
SIP port: 5560

Both test apps registered OK with our proxy. We configure 1 media proxy to be used with the LanScape SIP proxy.

We placed a call from extension 102 to 103. The proxy handled the call properly. The audio and media sessions were allocated on the media proxy as normal.

The problem is that we cannot figure out why you test app is responding to all INVITE requests consistently with 488 Not Acceptable Here SIP responses.

At this point in time, our SIP proxy + media proxy deployment is behaving as normal. Here is the SIP log fro the LanScape SIP proxy:

Code:

************* Log Opened (Oct 08 07:33:23) *************

<<<< (PROXY) RxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (From: 192.168.1.2:5560) <<<<

INVITE sip:103@192.168.1.2:5060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5560;branch=z9hG4bK-d8754z-4a435035ef34ca22-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:102@127.0.0.1:5560>

To: <sip:103@192.168.1.2:5060>

From: <sip:102@192.168.1.2:5060>;tag=ab42b812

Call-ID: MGM2NDA1ODZmYTdkMTU3ODM0MGQzMWI3MDUyODViY2Q.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, NOTIFY, REFER, REGISTER, SUBSCRIBE, MESSAGE, INFO

Content-Type: application/sdp

Supported: replaces

User-Agent: 3CXPhoneSystem

Content-Length: 303

v=0

o=- 9893648 9893648 IN IP4 192.168.1.2

s=http://www.portsip.com

c=IN IP4 192.168.1.2

t=0 0

m=audio 20382 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=ptime:20

a=fmtp:101 0-15

m=video 41522 RTP/AVP 125

a=fmtp:125 profile-level-id=42e015

a=rtpmap:125 H264/90000

>>>> (PROXY) TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (To: 192.168.1.2:5560) >>>>

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5560;

 branch=z9hG4bK-d8754z-4a435035ef34ca22-1---d8754z-;rport=5560;received=192.168.1.2

From: <sip:102@192.168.1.2:5060>;tag=ab42b812

To: <sip:103@192.168.1.2:5060>

Call-ID: MGM2NDA1ODZmYTdkMTU3ODM0MGQzMWI3MDUyODViY2Q.

CSeq: 1 INVITE

Server: LanScape Centrex Proxy/3.42.2.10 (www.LanScapeCorp.com)

Content-Length: 0

>>>> (PROXY) TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (To: 192.168.1.2:5560) >>>>

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 127.0.0.1:5560;branch=z9hG4bK-d8754z-4a435035ef34ca22-1---d8754z-;

 rport=5560;received=192.168.1.2

From: <sip:102@192.168.1.2:5060>;tag=ab42b812

To: <sip:103@192.168.1.2:5060>

Call-ID: MGM2NDA1ODZmYTdkMTU3ODM0MGQzMWI3MDUyODViY2Q.

CSeq: 1 INVITE

Proxy-Authenticate: Digest realm="ps", nonce="734481c1e7a139d58b699564eff216f4", 

 opaque="9596e8b548ba3c094cbb082faeb28dc3"

Server: LanScape Centrex Proxy/3.42.2.10 (www.LanScapeCorp.com)

Content-Length: 0

<<<< (PROXY) RxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (From: 192.168.1.2:5560) <<<<

ACK sip:103@192.168.1.2:5060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5560;branch=z9hG4bK-d8754z-4a435035ef34ca22-1---d8754z-;rport

Max-Forwards: 70

To: <sip:103@192.168.1.2:5060>

From: <sip:102@192.168.1.2:5060>;tag=ab42b812

Call-ID: MGM2NDA1ODZmYTdkMTU3ODM0MGQzMWI3MDUyODViY2Q.

CSeq: 1 ACK

Content-Length: 0

<<<< (PROXY) RxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (From: 192.168.1.2:5560) <<<<

INVITE sip:103@192.168.1.2:5060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5560;branch=z9hG4bK-d8754z-d34aec66c8769953-1---d8754z-;rport

Max-Forwards: 70

Contact: <sip:102@127.0.0.1:5560>

To: <sip:103@192.168.1.2:5060>

From: <sip:102@192.168.1.2:5060>;tag=ab42b812

Call-ID: MGM2NDA1ODZmYTdkMTU3ODM0MGQzMWI3MDUyODViY2Q.

CSeq: 2 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, NOTIFY, REFER, REGISTER, SUBSCRIBE, MESSAGE, INFO

Content-Type: application/sdp

Proxy-Authorization: Digest username="102",realm="ps",nonce="734481c1e7a139d58b699564eff216f4",

 uri="sip:103@192.168.1.2:5060",response="4ec7d911842dacfcba8c1ac5cb75620b",

 algorithm=MD5,opaque="9596e8b548ba3c094cbb082faeb28dc3"

Supported: replaces

User-Agent: 3CXPhoneSystem

Content-Length: 303

v=0

o=- 9893648 9893648 IN IP4 192.168.1.2

s=http://www.portsip.com

c=IN IP4 192.168.1.2

t=0 0

m=audio 20382 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=ptime:20

a=fmtp:101 0-15

m=video 41522 RTP/AVP 125

a=fmtp:125 profile-level-id=42e015

a=rtpmap:125 H264/90000

>>>> (PROXY) TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (To: 192.168.1.2:5560) >>>>

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 127.0.0.1:5560;

 branch=z9hG4bK-d8754z-d34aec66c8769953-1---d8754z-;rport=5560;received=192.168.1.2

From: <sip:102@192.168.1.2:5060>;tag=ab42b812

To: <sip:103@192.168.1.2:5060>

Call-ID: MGM2NDA1ODZmYTdkMTU3ODM0MGQzMWI3MDUyODViY2Q.

CSeq: 2 INVITE

Server: LanScape Centrex Proxy/3.42.2.10 (www.LanScapeCorp.com)

Content-Length: 0

>>>> (PROXY) TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (To: 192.168.1.80:5560) >>>>

INVITE sip:103@192.168.1.80:5560 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKf8997199b4633f42211c1a4e30594e39d.0

Via: SIP/2.0/UDP 127.0.0.1:5560;

 branch=z9hG4bK-d8754z-d34aec66c8769953-1---d8754z-;rport=5560;received=192.168.1.2

Record-Route: <sip:192.168.1.2:5060;lr>

From: <sip:102@192.168.1.2:5060>;tag=ab42b812

To: <sip:103@192.168.1.2:5060>

Call-ID: MGM2NDA1ODZmYTdkMTU3ODM0MGQzMWI3MDUyODViY2Q.

CSeq: 2 INVITE

Contact: <sip:102@192.168.1.2:5560>

max-forwards: 69

supported: replaces

Server: LanScape Centrex Proxy/3.42.2.10 (www.LanScapeCorp.com)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, NOTIFY, REFER, REGISTER, SUBSCRIBE, MESSAGE, INFO

Content-Type: application/sdp

Content-Length:   303

v=0

o=- 9893648 9893648 IN IP4 192.168.1.2

s=http://www.portsip.com

c=IN IP4 192.168.1.2

t=0 0

m=audio 16002 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=ptime:20

a=fmtp:101 0-15

m=video 16130 RTP/AVP 125

a=fmtp:125 profile-level-id=42e015

a=rtpmap:125 H264/90000

<<<< (PROXY) RxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (From: 192.168.1.80:5560) <<<<

SIP/2.0 488 Not Acceptable Here

Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKf8997199b4633f42211c1a4e30594e39d.0

Via: SIP/2.0/UDP 127.0.0.1:5560;

 branch=z9hG4bK-d8754z-d34aec66c8769953-1---d8754z-;rport=5560;received=192.168.1.2

To: <sip:103@192.168.1.2:5060>;tag=4f690b01

From: <sip:102@192.168.1.2:5060>;tag=ab42b812

Call-ID: MGM2NDA1ODZmYTdkMTU3ODM0MGQzMWI3MDUyODViY2Q.

CSeq: 2 INVITE

User-Agent: 3CXPhoneSystem

Content-Length: 0

>>>> (PROXY) TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (To: 192.168.1.80:5560) >>>>

ACK sip:103@192.168.1.80:5560 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bKf8997199b4633f42211c1a4e30594e39d.0

From: <sip:102@192.168.1.2:5060>;tag=ab42b812

To: <sip:103@192.168.1.2:5060>;tag=4f690b01

Call-ID: MGM2NDA1ODZmYTdkMTU3ODM0MGQzMWI3MDUyODViY2Q.

CSeq: 2 ACK

Server: LanScape Centrex Proxy/3.42.2.10 (www.LanScapeCorp.com)

Content-Length: 0

>>>> (PROXY) TxTxTxTxTxTxTxTxTxTxTxTxTxTxTx (To: 192.168.1.2:5560) >>>>

SIP/2.0 488 Not Acceptable Here

Via: SIP/2.0/UDP 127.0.0.1:5560;branch=z9hG4bK-d8754z-d34aec66c8769953-1---d8754z-;rport=5560;received=192.168.1.2

From: <sip:102@192.168.1.2:5060>;tag=ab42b812

To: <sip:103@192.168.1.2:5060>;tag=4f690b01

Call-ID: MGM2NDA1ODZmYTdkMTU3ODM0MGQzMWI3MDUyODViY2Q.

CSeq: 2 INVITE

Proxy-Authenticate: Digest realm="ps", nonce="734481c1e7a139d58b699564eff216f4", 

 opaque="9596e8b548ba3c094cbb082faeb28dc3"

Server: LanScape Centrex Proxy/3.42.2.10 (www.LanScapeCorp.com)

Content-Length: 0

<<<< (PROXY) RxRxRxRxRxRxRxRxRxRxRxRxRxRxRx (From: 192.168.1.2:5560) <<<<

ACK sip:103@192.168.1.2:5060 SIP/2.0

Via: SIP/2.0/UDP 127.0.0.1:5560;branch=z9hG4bK-d8754z-d34aec66c8769953-1---d8754z-;rport

Max-Forwards: 70

To: <sip:103@192.168.1.2:5060>;tag=4f690b01

From: <sip:102@192.168.1.2:5060>;tag=ab42b812

Call-ID: MGM2NDA1ODZmYTdkMTU3ODM0MGQzMWI3MDUyODViY2Q.

CSeq: 2 ACK

Content-Length: 0

************* Log Closed (Oct 08 07:34:04) *************

We tried all conceivable audio codec and video codec combinations. All with the same 488 result.

If you can help us understand your test app better, maybe we can run more tests and get your audio/video apps to connect. At this time, the problem does not appear to be our proxies. Out proxies will handle all the audio code types and video codec types your test apps support.

Note: We tried calls using other SIP UAs and our LanScape VOIP Media Engine based apps using only audio and we always get the same 488 Not Acceptable Here SIP responses from your test app code. We do not know why.

We would love to dig into this further with you but we would have to charge you our standard shop hourly rate for pre-sales support.

If you can tell us why your app always responds with 488 Not Acceptable Here SIP responses, we will try more tests.

Let us know what you want to do.

Support

khurram
Intermediate

Joined: October 06 2008
Location: Pakistan
Posts: 5

Posted: October 10 2008 at 4:23am \| IP Logged

According to my knowledge about SIP RFC and understanding.

"The user's agent was contacted successfully but some aspects of the session description such as the requested media, bandwidth, or addressing style were not acceptable.

A 606 (Not Acceptable) response means that the user wishes to communicate, but cannot adequately support the session described. The 606 (Not Acceptable) response MAY contain a list of reasons in a Warning header field describing why the session described cannot be supported. Warning reason codes are listed in Section 20.43."

its means that proxy is not able to understand the SDP Packet which our sdk is sending.. please verify it..

Regards

support
Administrator

Joined: January 26 2005
Location: United States
Posts: 1666

Posted: October 10 2008 at 8:43am \| IP Logged

Hi khurram,

We have no clued what you are talking about. Why are you discussing SIP 606 responses???? Whaaaaaa????

You are running a trial version of the PortSIP SDK. We cannot get your sample to work at all. We have configured a SIP test UA we use here, It will accept every type of call audio, video, you name it. When we test your PortSIP sample app against our call endpoint, your app constantly fails. This is not brain surgery we know what we are doing and there is something wrong with your sample app and your PortSIP SDK.

We suspect your trial license is no good anymore. We completely have the capability to help you but we cant afford to guess at what you are trying to do and waste time with expired trial software. We have tested your sample PortSIP app using other SIP test endpoints and the problem you are having has nothing to do with our SIP proxy products.

Here is what you should do:

1)
Get a real license to the PortSIP SDK not a trial license.

2)
Take two instances of your sample app. One running on machine A and another one running on Machine B. Make sure your sample apps can call each other without the use of a proxy. If we cant get this simple scenario to work, then we are screwed. Stop immediately, go home, drink a beer and forget that PortSIP ever existed.

3)
Once you get step 2 running, we can then add our LanScape proxy products to the mix. Looking at the SIP from the PortSIP sample app, our proxies will work without error. Calls consisting of audio and video will work. Our proxies support up to 128 media sessions per call. That is, 128 total media sessions per call any mix of audio and video is OK.

Final Comment:
Something is not right with your PortSIP SDK. We even tried audio calls only and as soon as your sample app makes an outgoing call and our SIP test device answers the call (uLaw codec only), your sample app terminates the call immediately by issuing a SIP BYE request. Also, anytime we try to call your sample app, it always responds with a 488 Not acceptable Here response. We have no clue why

If you really need help, we have everything you need to get you up and running. Consider entering into a short term paid support period with us. If you do that, we will get you going.

Support

khurram
Intermediate

Joined: October 06 2008
Location: Pakistan
Posts: 5

Posted: October 13 2008 at 12:46am \| IP Logged

Final Comment:
Something is not right with your PortSIP SDK. We even tried audio calls only and as soon as your sample app makes an outgoing call and our SIP test device answers the call (uLaw codec only), your sample app terminates the call immediately by issuing a SIP BYE request. Also, anytime we try to call your sample app, it always responds with a 488 Not acceptable Here response. We have no clue why

This is what the answer i need to hear from your side. i want to know actual problem which u people identified. thank you for this...

i will come back soon after getting the sdk which supports video and audio. if you have any sdk for C#. please refer me .. Thank you.

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