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ASK THE EXPERT - SIP TRUNKING

Welcome to the Cisco Networking Professionals Ask the Expert conversation. This is an opportunity to learn more about SIP Trunking with Cisco expert Christina Hattingh. Christina is a technical marketing engineer in the branch office Unified Communications group at Cisco, focusing on unified communications technologies and network deployments on the Cisco 2800 and 3800 Series Integrated Services Routers. Unified communications technologies integrated in Cisco IOS Software offer numerous communications services, such as voice and video gateways, call control, network services, and applications. In this role, Hattingh helps guide development projects, trains Cisco sales staff and Cisco resale partners on new Cisco IOS-based voice technologies, and advises customers on unified communications network deployment and design. Hattingh has been with Cisco Systems since 1998 and holds a graduate degree in computer science and mathematical statistics.

Remember to use the rating system to let Christina know if you have received an adequate response.

Christina might not be able to answer each question due to the volume expected during this event. Our moderators will post many of the unanswered questions in other discussion forums shortly after the event. This event lasts through January 15, 2010. Visit this forum often to view responses to your questions and the questions of other community members

27 REPLIES
New Member

Re: ASK THE EXPERT - SIP TRUNKING

Hi,

I have one CUCM 5.0.4 server with one SIP trunk to a 3rd Party PABX (non-cisco). Voice Calls between them are working fine but faxes are not working.

In Cisco Side i  have faxe machines conneted to ATA's 186 (SCCP) and in non-cisco side i also have 3rd Party ATA's connected to 3rd Party PABX.

Fax machine rings but connection get dropped and fax are not transmited.

Is fax over sip trunk supported in these scenario?

Are T.38 supported in this  version of CUCM (5.0.4)? Or what fax protocol should be used?

Any suggestions for configuration setup?

Thanks.

Best Regards,

MC

Cisco Employee

Re: ASK THE EXPERT - SIP TRUNKING

Cisco ATAs only support a Cisco proprietary version of passthrough for fax calls and is therefore not compatible with third party devices. In situations like this, the common solution is to always use a G.711 voice codec between the ATA endpoints connected to fax machines in what Cisco terms "fax of G.711 voice".

T.38 is supported for SIP in CUCM 5.x but the ATA186 does not support T.38.I am not sure how many fax endpoints you have but the Cisco VG202 or VG204 may be an option that offers full T.38 support in an ATA-size form factor.

Regards,

David

New Member

Re: ASK THE EXPERT - SIP TRUNKING

So let me see if i understood If have fax machine connected on VG202 (that supports T.38) in CUCM side and if have 3rd party ATA (that supports T.38) connected to fax machine  in 3rd Party PABX i could sent faxes sucessfully via SIP Trunk beetween CUCM and 3rd party PABX?

Is these the only solution?

PS:Fax between CUCM and 3rd party PABX are working when going trough PSTN.

Thanks

Best Regards,

MC

Cisco Employee

Re: ASK THE EXPERT - SIP TRUNKING

Yes, that should work barring any T.38 interoperability issues that occasionally crop up between vendors. From the CUCM 5.x perspective, T.38 is only supported using SIP and H.323. So, the VG202 would need to be communicating with CUCM using one of these two call control protocols.

From a solution perspective, you are looking at two choices:

1) Use existing setup and run all fax calls over G.711 - fax over G.711 voice

2) Replace Cisco ATAs with VG202 or similar device supporting standards-based T.38 fax relay and use T.38 to transport the fax calls. Of course, the third party devices on the other side of the SIP trunk must also support T.38.

There are also fax passthrough solutions but because of the proprietary nature of Cisco's modem passthrough switchover and the lack of support in CUCM for fax pass-through using call control protocol, these are not valid options in your scenario.

Regards,

David

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Can a 3845 be configured to use MGCP and also run CUBE?

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Yes, it can. In general Voice GW functions (MGCP, SCCP, SIP, H.323) can be used/collocated with CUBE.

CH

Hall of Fame Super Gold

Re: ASK THE EXPERT - SIP TRUNKING

Hello Christina.

Are there plans to allow outgoing authentication for multiple trunks ?

That is, the ability of using multiple SIP username/password to the same or different servers.

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Yes, there is a Multiple Registrar (up to 6) feature in CUBE 1.4 (15.0.1XA available now on selected platforms, and 15.1.1T soon with remaining platforms).

http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-multi-registrars.html

CH

Hall of Fame Super Gold

Re: ASK THE EXPERT - SIP TRUNKING

Chistina,

that is good, but unless I misunderstand the feature, it is not what my customers want.

They want to be able to place outgoing calls to ITSP "abc" using a given set of credentials, and at the same time place calls to ITSP "def" using another set of credentials. This, indipendetly from any registration to ITSP.

It is possible to do that using the feature you linked? If not, will be possible at some given time ?

Thanks, Paolo.

New Member

Re: ASK THE EXPERT - SIP TRUNKING

So if I understand your setup correctly, you do not want CUBE to register. But you want CUBE to respond to INVITE challenges when CUBE initiates outgoing calls towards the SP softswitch.

If so, then it would appear you have to use the "authentication" CLI, as opposed to the "credentials" CLI. The "credentials" CLI is used for registration. "authentication" CLI is used for responding to challenges to either Registration or INVITE requests. So if CUBE sends and INVITE to softswitch-A, and softswitch-A challenges it, then CUBE will respond with the credentials in the matching "authentication" CLI. Matching means the realm quoted in the challenge from softswitch-A must match the realm in the "authentication" CLI on CUBE, from which it then extracts the username/passwd to repond to softswitch-A's challenge.

The "authentication" CLI under sip-ua is not a new capability in IOS, it's been there a while:

http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_a1.html#wp1631772

However, my understanding is that they expanded the number of these statements that you can have under the sip-ua configuration in 15.0.1XA. You would need one statement for each different ITSP softswitch (realm) you want CUBE to respond to with different credentials. Even older version of IOS supported multiple statements, so you should be able to do this w/o upgrading your IOS.

CH

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Greetings,

My client has a SIP Trunk in one of their sites.  They had mulitple problems with DTMF for outbound and Inbound to IVR.  Currently, they're running CUCM6.1 and UCCX 7.x  The Gateway where the SIP trunk resides is 2801.  The carrier is using BTS (Cisco Soft switch).  After opening Tickets and TAC, they came to the cnclusion that they had to create CME 1 ephone 1 ephone-dn and register a Transcoder under Telephoney Service.  They also have another transcoder registered to the CUCM (2nd transcoders).  Plus the SIP trunk between CUCM and the Gateway is using MTP (without it nothing works).  I failed to mention that this is a Centralized deployment.  Now, I did some cleanup on Dialpeers and I can dial out many IVR's and it works perfectly except certain one's that dtmf doesn't work.  Without the Transcoders nothing works.

Do we need two transcoders and MTP to get a SIP Trunk running?  If DTMF works for many IVR's and not others what's the best path to troubleshoot this problem?  Any ideas or suggestions??  I would like to start by removing transcoders to see which one works and which one that doesn't.  Thanks in advance.

DTMF type = RTP-NTE (I will try Sip Notify then I will remove the dtmf from Dial-peers to test)

Payload = 101 (confirmed by the carrier).

New Member

Re: ASK THE EXPERT - SIP TRUNKING

I  assume the 2801 is a CUBE (SIP-SIP IPIPGW)?

It sounds like this isn't configured optimally. Why is there an MTP in the path? If it is to enable SIP early offer from CUCM, then an easier way might be to have CUCM do Delay Offer and have CUBE do the DO-EO conversion for the SP SIP trunk.

Further, you don't need xcoders for DTMF interop. Xcoders are only required if you're changing the codec, so it seems like there is some more fundamental mismatch that inserting a xcoder just happen to mask. What is the DTMF method used/preferred by each side? Why (what error code) do calls fail with when trying to negotiate DTMF and there is no xcoder in the path? Is there perhaps a payload type mismatch for RFC 2833 DMTF from the two sides? If so, a new feature in CUBE 1.4 to do payload type conversion/interworking might be helpful to your setup.

http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-sipsip_ps5640_TSD_Products_Configuration_Guide_Chapter.html#wp1403792

CH

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Hi,

Thanks for the reply.  New info, the carrier only Supports In-Band and NOT OOB.  That might explain the use of Transcoders.  Anyway, Is there a way to send DTMF In-Band instead OOB?  I'm not sure how to do that.

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Ah yes, now that makes a difference. For pure inband G.711 tone detection you will need DSPs in the path and the way to do that is with a xcoding configuration, so you have the right piecepart for this. For DTMF relay conversion (from one type/protocol to another), no DSPs/xcoding is needed).

When no DTMF relay is configured on a dial-peer, the default operation is inband. The config below will do inband to RFC2833 conversion (with support from the DSPs which are part of the xcoding config.

CH

dial-peer voice xxx voip
description ** incoming calls from SP ** <<<<=== Incoming Dialpeer
session protocol sipv2
incoming called-number (enterprise DID range) 
codec g711ulaw <== No DTMF is configured, which means raw inband DTMF
no vad
!
dial-peer voice yyy voip
destination-pattern (enterprise DID range) <<<<=== Outgoing Dialpeer
session target ipv4:CUCM's-IP-address
dtmf-relay rtp-nte digit-drop <== RFC2833 (rtp-nte) with digit-drop is configured for the leg to CUCM
codec g711ulaw <== Assuming you want G.711 all throughout, else this can be G.729 as well, the same xcoder will be used
no vad

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Hi,

Registering transcoding resources locally fixed all my DTMF-related problems. Now it works as required.

A couple of 'hidden' commands benefited me. I had a SRST instance on the troublesome router, under which (the call-manager-fallback section) the dsp farms are registered. The hidden commands which did the trick for me were:

sdspfarm units 1

sdspfarm sessions 5 (or any other number)

No documentation I found told me of this, I was just comparing transcoding config to another router with CCME on it (under telephony-service, these commands are not hidden) and everything fell into place.

New Member

Re: ASK THE EXPERT - SIP TRUNKING

I have another SIP DTMF question. I've been reading the docs but I haven't found anything that matches my scenario yet. I have an Advanced SIP device running (AlarmPoint) on CUCM 7.x. We are using MGCP as the signaling protocol on the VGs. We want to have AlarmPoint answer inbound calls for our NOC agents and then blind transfer that call to a conference bridge 800# back out the PRIs. Yes this will consume 2 PRI channels per call to AlarmPoint but we have 2 PRIs and utilization isn't heavy enough to consume all of the channels. Only problem is I can't figure out the correct method for delivering DTMF. I was thinking RTP-NTE. or SIP Notify. Any suggestions?

New Member

Re: ASK THE EXPERT - SIP TRUNKING

The CUCM folks advise that you can use either. CUCM supports RFC2833 on MGCP as well as SIP (it's default on SIP, but not on MGCP). CUCM also supports out-of-band DTMF methods on both MGCP and SIP (this is default on MGCP, and a SIP Notify config on SIP).

As long as you have the same DTMF method on both sides it should work w/o requiring an MTP. If you use different methods on MGCP vs. SIP it can still work, but will require an MTP.

So it really depends on what your SIP application wants to see. RFC2833 e2e may be the easiest option.

CH

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Inband tone conversion to DTMF relay is supported only for G.711 (on the inband side), and RFC2833 (on the DTMF relay side). not for the OOB relay methods such as SIP Notify, KPML, H.323-alpha/signal etc.

CH

New Member

Re: ASK THE EXPERT - SIP TRUNKING

I have a cisco 7911 phone with sip firmware connected to a third party sip server. We can call in and out from the phone. However, if this phone is calling to site that had auto attendant that required to key in number(please press 1 for sale, 2 for support etc), number press is not taken in.

We try using another sip device(linksys phone) that has a option of chosing what type of dtmf to use. When we use inband, it work fine , but when we chose to use OOB the same problem happen.

We found out that, by default this 7911 sip phone uses OOB dtmf in this sip cnf.xml. Is there a way to change dtmf to inband?  Thank you

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New Member

Re: ASK THE EXPERT - SIP TRUNKING

The CUCM/phone group advises that the 7911 with a SIP load is only supported with Cisco call control. Nevertheless, the DTMF method that gets selected for a call is a result of the call control negotiation, rather than a phone setting. So you will need to change 3rd party SIP server to negotiate/force RFC2833 for these phones to make DTMF work in your scenario. The phone supports both methods and will offer both methods in its SDP. Which one gets selected depends on what the remote side offers and what call control selects.

CH

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Is there any advantage/disadvantage of using SIP trunking versus using a Intercluster Trunk (ICT) between CUCM clusters?

Thanks

New Member

Re: ASK THE EXPERT - SIP TRUNKING

CUCM ICT trunks can be either H.323 or SIP.

The main advantage of using traditional H.323-based ICTs is that it supports Q.SIG tunneling when interconnecting CUCM with existing Q.SIG PBXs allowing for the support of various supplementary services between CUCM and PBX endpoints. H.323 ICT also has built in destination redundancy as you can define up to three destination addresses.

SIP ICT has certain advantages as it supports of a larger set of codecs, Enhanced MWI, etc.

CH

Hall of Fame Super Gold

Re: ASK THE EXPERT - SIP TRUNKING

Any answer to my question above ?

Chistina,

that is good, but unless I misunderstand the feature, it is not what my customers want.

Theywant to be able to place outgoing calls to ITSP "foo" using a given setof credentials, and at the same time place calls to ITSP "bar" usinganother set of credentials. This, independently from any registration to ITSP.

It is possible to do that using the feature you linked? If not, will be possible at some given time ?

Thanks, Paolo.

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Yes, I posted an answer this morning to your original question. Copied here as well:

So if I understand your setup correctly, you do not want CUBE to register. But you want CUBE to respond to INVITE challenges when CUBE initiates outgoing calls towards the SP softswitch.

If so, then it would appear you have to use the "authentication" CLI, as opposed to the "credentials" CLI. The "credentials" CLI is used for registration. "authentication" CLI is used for responding to challenges to either Registration or INVITE requests. So if CUBE sends and INVITE to softswitch-A, and softswitch-A challenges it, then CUBE will respond with the credentials in the matching "authentication" CLI. Matching means the realm quoted in the challenge from softswitch-A must match the realm in the "authentication" CLI on CUBE, from which it then extracts the username/passwd to repond to softswitch-A's challenge.

The "authentication" CLI under sip-ua is not a new capability in IOS, it's been there a while:

http://www.cisco.com/en/US/docs/ios/voice/command/reference/vr_a1.html#wp1631772

However, my understanding is that they expanded the number of these statements that you can have under the sip-ua configuration in 15.0.1XA. You would need one statement for each different ITSP softswitch (realm) you want CUBE to respond to with different credentials. Even older version of IOS supported multiple statements, so you should be able to do this w/o upgrading your IOS.

CH

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Greetings,

What's the recommended billing solution for CUBE? Who has any working solution recommendation?

Thanks a lot!

Joe

CCIE#6909

New Member

Re: ASK THE EXPERT - SIP TRUNKING

CUBE supports CDR records via AAA/RADIUS or SNMP Call History MIB, just like the IOS TDM GWs, CME and SRST do.

I don't believe Cisco has any inhouse billing solutions, these are provided via the Cisco partner community. I'm not familiar with these solutions, but they would be the same as they are for the TDM GWs. MindCTI is a longtime general Cisco billing partner, but I don't know the specifics of their products or solutions.

CH

New Member

Re: ASK THE EXPERT - SIP TRUNKING

Hello!

Whether Linksys SPA-(2102|8000) can restore voice  session after a fax t.38  sending?

I tried to send a fax (t.38) through the softswitch and directly. Fax pass, but voice session is not restored.
From debug it is visible that SPAs do not send rInvite.
If rInvite comes from the softswitch, SPAs sends .

I used last firmware.

Thanks.

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