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Beginner

Issues with SIP when registered to VCSE

Hi Guys,

VCSE and VCSC are both on X5.2. When a movi client is registered to the VCSE and we try to dial into the MPS800 which is registered to the vcsc the particpants recieve no video or audio. However when the same call is made using H323 the call fucntions fine. Is there any way to force the vcse to do a h323 through the firewall into the expressway.

thanks

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1 ACCEPTED SOLUTION

Accepted Solutions
Highlighted
Cisco Employee

Issues with SIP when registered to VCSE

No, you must configure VCS-C address as SIP server address on MPS.

Otherwise static conference with SIP and single dial in number with SIP will not register on VCS-C.

For VCS to send call to MPS when receive personal conference URL from other, you need to configure as blow (just repeat what I mention earlier).

====================================================================================

[MPS]

- SIP Server: VCS-C IP address

- Default transport protocol: TCP (you may use TLS if you want)

- Configure personal conference URL (with specific format so may easily configure search rule on VCS forwarding it to MPS not overlap with other Endpoint/SIP UA alias).

[VCS]

- Neighbor zone

  Type: Neighbor

  H323 Mode: Off

  SIP: Mode: on

  SIP Port: 5060 (assume MPS transport protocol configured as TCP)

  SIP Transport: TCP (assume MPS transport protocol configured as TCP)

  Location Peer 1 address: MPS SC IP address

Zone profile: Default

- Search rule

  Mode: Alias pattern match

  Pattern type: Regex

  Pattern string: 98.*@domain.com (assume personal conference alias is 98xxxxx@domain.com)

  Pattern behavior: Leave

  Target: MPS-SIP-Neighbor (this is SIP neighbor zone created and point to MPS)

====================================================================================

With above configuration, VCS-C forward any SIP call with URL 98xxxx@domain.com forward to MPS.

I'd recommend you first make a test call within VCS-C (not call from VCS-E) to verify call establishment.

View solution in original post

33 REPLIES 33
Highlighted
Beginner

Issues with SIP when registered to VCSE

this is the call that fails (sip to sip from movi to mps conference)

Jun 3 20:20:04tvcs: Event="

Call Rejected

" Service="

SIP

" Src-ip="

86.130.55.41

" Src-port="

49220

" Src-alias-type="

SIP

" Src-alias="

sip:adeel.ahmed@callpresence.com

" Dst-alias-type="

SIP

" Dst-alias="

sip:171011@callpresence.com

" Call-serial-number="

1f376b42-adb1-11e1-9ceb-0010f316e148

" Tag="

1f376c96-adb1-11e1-b201-0010f316e148

" Detail="

Unsupported URI Scheme

" Protocol="

TLS

" Response-code="

416

" Level="

1

" UTCTime="

2012-06-03 19:20:04,297

this is a call that connects (sip to sip from movi to mps auto attendent)

"

Call Connected

" Service="

SIP

" Src-ip="

87.127.38.2

" Src-port="

51565

" Src-alias-type="

SIP

" Src-alias="

sip:joel.ettienne@callpresence.com

" Dst-alias-type="

SIP

" Dst-alias="

sip:linkmps@callpresence.com

" Call-serial-number="

1a17b9c2-ad08-11e1-9cc2-0010f316e148

" Tag="

1a17bb2a-ad08-11e1-afbb-0010f316e148

" Protocol="

TLS

" Call-routed="

YES

" Level="

1

" UTCTime="

2012-06-02 23:10:10,895"

this is a call that connects (h323 to h323  from EP to mps conference)

"

Call Connected

" Service="

H323

" Src-ip="

87.127.38.2

" Src-port="

3232

" Dst-alias-type="

E164

" Dst-alias="

171011

" Call-serial-number="

f7b6443a-ad7e-11e1-809e-0010f316e148

" Tag="

f7b64598-ad7e-11e1-9e73-0010f316e148

" Protocol="

TCP

" Level="

1

" UTCTime="

2012-06-03 13:21:03,574

"

Highlighted
Cisco Employee

Issues with SIP when registered to VCSE

Hi Shazad,

If you check the call flow for the H.323 and SIP you will see the difference. In case of H.323 the dest alias is 171011, however in case of SIP the dest alias is 171011@callpresence.com.

From the logs it also says error code 416 which is unsupported URI scheme..!!may be you need a transform in case of SIP calls which will strip the domain and forward the calls to MPS.

What about SIP services on MPS? is it on?

Thanks

Alok

Highlighted
Cisco Employee

Issues with SIP when registered to VCSE

Sounds like possible with firewall that deny Ack message on invite therefore only one way media on call.

> Is there any way to force the vcse to do a h323 through the firewall into the expressway.

Yes, by enable only H.323 traversal link between VCS-E and VCS-C (simply create traversal zone just enable H323 but disable SIP) and enable interworking (interworking -> on), then call between VCS-E and VCS-C will negotiate over H.323.

VCS-E and VCS-C will handle SIP call is call receive from/to SIP UA (but still use H323 call method over traversal link).

Highlighted
Beginner

Issues with SIP when registered to VCSE

Hi Both,

Just for info the MPS is registering fine SIP also we have tried disabling SIP on the traversal to just use H.323 but this breaks the MOVI registrations or stops anybody trying to register there MOVI client.

I think the search rules we have in place on the expressway are ok to pass movi call through to the traversal zone

many thanks

Shaz

Highlighted
Rising star

Re: Issues with SIP when registered to VCSE

Shazad,

is '171011@callpresence.com' the exact SIP AOR which the MPS registered to the VCS-C for the conference you are attempting to dial into?

Do you have any transforms on VCS-E or VCS-C which strips away the domain portion of aliases such as this?

Can you post screenshots of the search rules and MPS registrations on VCS-C?

Thanks,

Andreas

Highlighted
Beginner

Issues with SIP when registered to VCSE

Hi andreas,

VCSC:

VCSE Transforms:

VCSC registrations:

URI's we can dial but not but not 11111 or 33333 aliases etc.

Thanks

Shaz

Highlighted
Rising star

Issues with SIP when registered to VCSE

Shazad,

what is the SIP AOR of the MPS conference you want to dial into?

I wouldn't recommend having any transforms stripping away the domain portion of URI's since that will effectively break SIP URI dialling.

Do you wish to dial into the MPS conference on SIP, so that the call remains in SIP the entire way, or do you wish for the call to be interworked to H323 on the VCS-C before it is sent to the MPS?

Highlighted
Beginner

Issues with SIP when registered to VCSE

Hi Andreas,

The plan is to have users dial conferences starting with 171*** to reach there respective VMR ideally we would like this internetworked from SIP-H.323 to the MPS.

what would you see best?

Also i have disabled the trasnforms on the expressway and this is now working on 171011 and 11111, SIP all th way through.

Thanks

Shaz

Highlighted
Beginner

Issues with SIP when registered to VCSE

Guys,

All your help so far is appreciated, below is the behaviour were experiencing with any but used movi as an example below:

When you dial into a personal conference with movi registered to the expressway:

No video or audio on h323 or sip

When you dial into a static conference with movi registered to the expressway:

Audio and video on h323 or sip

Highlighted
Rising star

Issues with SIP when registered to VCSE

Shazad,

it would be quite difficult to pinpoint the cause of this media issue without collecting logs from your VCS's and MPS.

Are there any differences in configuration for the personal and static conferences on the MPS with regards to codecs, layouts etc?

Since you mention running X5.2 on your VCS's, it could very well be that you will have better luck with upgrading your VCS's to X7.1 and the MPS to J4.6 of not already running this.

If upgrading does not change anything, I would recommend you raise a case with TAC to troubleshoot further.

- Andreas

Highlighted
Beginner

Issues with SIP when registered to VCSE

Big thanks for all your help so far guys, u guy are ace.

I have a question, is there any way we can translate any movi sip calls to h323? This is because we are having issues with sip to sip calls to the mps.

Highlighted
Cisco Employee

Issues with SIP when registered to VCSE

VCS support interworking call converting SIP to H.323 and H.323 to SIP.

Configuration is available from “VCS Configuration > Protocols > Interworking” on Web GUI.

Off: the VCS will not act as a SIP-H.323 gateway.
Registered only: the VCS acts as a SIP-H.323 gateway but only if at least one of the endpoints is locally registered.
On: the VCS acts as SIP-H.323 gateway regardless of whether the endpoints are locally registered.

Default is registered only.

Highlighted
Beginner

Issues with SIP when registered to VCSE

Hi Tomonori,

thanks fro the quick reply. I am aware of the functionality you have described above. As we are able to dial in from a h323 EP registered to our vcse directly into our mps800 without issue, we would therefore like the vcse to translate any calls made from a sip movi client (registered to vcse) to h323 before it reaches the control on the opposite side of the firewall. we need to be able to do this via a transform or search rule that won't stop movi registrations from the public internet.

Highlighted
Cisco Employee

Issues with SIP when registered to VCSE

If interworking call feature enable in multiple VCS in call path, then interworking call will handle by VCS which closest to Endpoint that require H323-SIP translate.

For your environment (in default deployment), VCS-C will handle the interworking call.

There are few methods to force VCS-E to handle interworking call.

- Only enable H323 traversal link between VCS-C and VCS-E on traversal zone configuration

- Disable interworking feature on VCS-C

- Strip domain from URL for incoming call to MPS on VCS-E.
For example, assume E.164 alias on MPS is 2xxxx then have search rule (2.*)@domain.com -> replace -> \1, then VCS-E will send LRQ for 2xxxx which VCS-C will response LCF as H323 call.

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