CUCM - DSCP for TelePresence Calls Enterprise Parameter

Hi All,
There is an enterprise parameter in newer versions of CUCM called "DSCP for TelePresence Calls" that is separate from "DSCP for Video Calls".
Does anyone know what endpoints the former applies to? Is it just the traditional CTS endpoints (e.g CTS-3000), or is it all devices that include TelePresence in the name in CUCM for example Cisco TelePresence EX60?
Cheers,
Giles

It's only for immersive room endpoints (e.g. CTS1300, 3000, TX9000, and the like).
Sorry for the delayed response, I just spotted this as part of an unrelated Google search and am answering incase someone else finds this in their search results.

Similar Messages

Dx650 - Just CUCM for video calling?

We have CUCM 10, running on a BE6K. We just ordered some video endpoints (DX650).
Will person to person video calling work for users on our CUCM cluster? I realize for external calling we need Cisco TelePresence Server/Cisco TelePresence Conductor/Expressway. But I assume for internal phone to phone video we don't need anything?
And CUWL standard will be sufficient?
Thanks, Bill

Point to point calling will work without any issues. You may need to have enhanced license as I believe it is seen as a telepresence device.

Unity Connection - Certificate from cucm no more trusted for encrypted calls after upgrade to 10.5(1)

Hello Support Community,
i have a strange problem:
after upgrading my cucm and unity connection from 9.1 to 10.5(1) enctrypted calls are no more working.
situation 1: CUCM is down, Subscriber is up: Encrypted call to Unity Connection work correctly
situation 2: CUCM is up: Encrypted Calls to Unity Connection not working.
i get the following Info in the log for the Connection Conversion Manager:
19:35:21.053 |15865,,,MiuGeneral,25,Invalid Certificate: Received Certificate -----BEGIN CERTIFICATE-----
MIID8zCCAtugAwIBAgIQc/fBdUz1Zdh4CXhcPqGVuDANBgkqhkiG9w0BAQsFADBw
MQswCQYDVQQGEwJERTELMAkGA1UEChMCSVQxGzAZBgNVBAsTEkhlbGxnYXRlIFRl
XD0oD9d5MQ==
-----END CERTIFICATE-----
doesn't match with stored Certificate: -----BEGIN CERTIFICATE-----
MIIC2DCCAkGgAwIBAgIIJWCm4bSdt+kwDQYJKoZIhvcNAQEFBQAw
-----END CERTIFICATE-----
so where does Unity Connection cache this certificate and how can i delete/replace it?
the cert shown in the logs is the one from cucm: ("CallManager"), i recreated it through cucm os administration, now i see the same error message on unity connection for the new recreated certificate.

Actually It doesn't. It says he's on a MacBook. I don't know all the different types of Macs. I was having a ton of problems with iChat. I opened DMZ to my computer, knocked down all firewalls etc and left everything exposed, still with bad results. A few weeks ago my power supply went out on my D-Link. I bought a linksys. Since I'd left all firewalls off I figured it couldn't be the router. I power cycled everything n the netork, still no luck. Today I bought a universal Power supply and started up my D-Link Router. Everything worked perfectly. My wifes computer - a laptop running Tiger worked fine with the Linksys and did my machine before the Leopard upgrade. Now that I've got the D-Link online everythings working.
Message was edited by: graphico
Message was edited by: graphico

How to define parameter ID for transaction call in query.

Hi folks,
In SQ01 I've made report assignment. So now buy double click from ALV table I'm calling a new transaction and to fill fields on first screen I need define parameters ID. So how can I do it in query?
Thanks.

hi,
try this snippet,
FORM user_command_form USING p_ucomm    LIKE sy-ucomm
                              p_selfield TYPE slis_selfield.
CASE p_ucomm.
    WHEN '&IC1'.        " SAP standard code for double-clicking
           SET PARAMETER ID 'XXX' FIELD p_selfield-value.
      CALL TRANSACTION 'YYYY'.
ENDCASE.
ENDFORM.                    "user_command_form
Hotspot and FM REUSE_ALV_GRID_DISPLAY not working

Call for participation: OASIS Enterprise Key Management Infrastructure TC

We would welcome your participation in this process. Thank you.
Arshad Noor
StrongAuth, Inc.
To: OASIS members & interested parties
A new OASIS technical committee is being formed. The OASIS Enterprise Key
Management Infrastructure (EKMI) Technical Committee has been proposed by the
members of OASIS listed below. The proposal, below, meets the requirements of
the OASIS TC Process [a]. The TC name, statement of purpose, scope, list of
deliverables, audience, and language specified in the proposal will constitute
the TC's official charter. Submissions of technology for consideration by the
TC, and the beginning of technical discussions, may occur no sooner than the
TC's first meeting.
This TC will operate under our 2005 IPR Policy. The eligibility
requirements for becoming a participant in the TC at the first meeting (see
details below) are that:
(a) you must be an employee of an OASIS member organization or an individual
member of OASIS;
(b) the OASIS member must sign the OASIS membership agreement [c];
(c) you must notify the TC chair of your intent to participate at least 15
days prior to the first meeting, which members may do by using the "Join this
TC" button on the TC's public page at [d]; and
(d) you must attend the first meeting of the TC, at the time and date fixed
below.
Of course, participants also may join the TC at a later time. OASIS and the TC
welcomes all interested parties.
Non-OASIS members who wish to participate may contact us about joining OASIS
[c]. In addition, the public may access the information resources maintained for
each TC: a mail list archive, document repository and public comments facility,
which will be linked from the TC's public home page at [d].
Please feel free to forward this announcement to any other appropriate lists.
OASIS is an open standards organization; we encourage your feedback.
Regards,
Mary
Mary P McRae
Manager of TC Administration, OASIS
email: mary.mcrae(AT)oasis-open.org
web: www.oasis-open.org
a) http://www.oasis-open.org/committees/process.php
b) http://www.oasis-open.org/who/intellectualproperty.php
c) See http://www.oasis-open.org/join/
d) http://www.oasis-open.org/committees/tc_home.php?wg_abbrev=ekmi
CALL FOR PARTICIPATION
OASIS Enterprise Key Management Infrastructure (EKMI) TC
Name
OASIS Enterprise Key Management Infrastructure (EKMI) TC
Statement of Purpose
Public Key Infrastructure (PKI) technology has been around for more than a
decade, and many companies have adopted it to solve specific problems in the
area of public-key cryptography. Public-key cryptography has been embedded in
some of the most popular tools -- web clients and servers, VPN clients and
servers, mail user agents, office productivity tools and many industry-specific
applications -- and underlies many mission-critical environments today.
Additionally, there are many commercial and open-source implementations of PKI
software products available in the market today. However, many companies across
the world have recognized that PKI by itself, is not a solution.
There is also the perception that most standards in PKI have already been
established by ISO and the PKIX (IETF), and most companies are in
operations-mode with their PKIs -- just using it, and adopting it to other
business uses within their organizations. Consequently, there is not much left
to architect and design in the PKI community.
Simultaneously, there is a new interest on the part of many companies in the
management of symmetric keys used for encrypting sensitive data in their
computing infrastructure. While symmetric keys have been traditionally managed
by applications doing their own encryption and decryption, there is no
architecture or protocol that provides for symmetric key management services
across applications, operating systems, databases, etc. While there are many
industry standards around protocols for the life-cycle management of asymmetric
(or public/private) keys -- PKCS10, PKCS7, CRMF, CMS, etc. -- however, there is
no standard that describes how applications may request similar life-cycle
services for symmetric keys, from a server and how public-key cryptography may
be used to provide such services.
Key management needs to be addressed by enterprises in its entirety -- for both
symmetric and asymmetric keys. While each type of technology will require
specific protocols, controls and management disciplines, there is sufficient
common ground in the discipline justifying the approach to look at
key-management as a whole, rather than in parts. Therefore, this TC will
address the following:
Scope
A) The TC will create use-case(s) that describe how and where
the protocols it intends to create, will be used;
B) The TC will define symmetric key management protocols,
including those for:
1. Requesting a new or existing symmetric key from a server;
2. Requesting policy information from a server related to caching of keys on the
client;
3. Sending a symmetric key to a requestor, based on a request;
4. Sending policy information to a requestor, based on a request;
5. Other protocol pairs as deemed necessary.
C) To ensure cross-implementation interoperability, the TC will create a test
suite (as described under 'Deliverables' below) that will allow different
implementations of this protocol to be certified against the OASIS standard
(when ratified);
D) The TC will provide guidance on how a symmetric key-management infrastructure
may be secured using asymmetric keys, using secure and generally accepted
practices;
E) Where appropriate, and in conjunction with other standards organizations that
focus on disciplines outside the purview of OASIS, the TC will provide input on
how such enterprise key-management infrastructures may be managed, operated and
audited;
F) The TC may conduct other activities that educate users about, and promote,
securing sensitive data with appropriate cryptography, and the use of proper
key-management techniques and disciplines to ensure appropriate protection of
the infrastructure.
List of Deliverables
1. XSchema Definitions (XSD) of the request and response protocols (by August
2007) 2. A Test Suite of conformance clauses and sample transmitted keys and
content that allows for clients and servers to be tested for conformance to the
defined protocol (by December 2007)
3. Documentation that explains the communication protocol (by August 2007)
4. Documentation that provides guidelines for how an EKMI may be built,
operated, secured and audited (by December 2007)
5. Resources that promote enterprise-level key-management: white papers,
seminars, samples, and information for developer and public use. (beginning
August 2007, continuing at least through 2008)
Anticipated Audiences:
Any company or organization that has a need for managing cryptographic keys
across applications, databases, operating systems and devices, yet desires
centralized policy-driven management of all cryptographic keys in the
enterprise. Retail, health-care, government, education, finance - every industry
has a need to protect the confidentiality of sensitive data. The TC's
deliverables will provide an industry standard for protecting sensitive
information across these, and other, industries.
Security services vendors and integrators should be able to fulfill their use
cases with the TC's key management methodologies.
Members of the OASIS PKI TC should be very interested in this new TC, since the
goals of this TC potentially may fulfill some of the goals in the charter of the
PKI TC.
Language:
English
IPR Policy:
Royalty Free on Limited Terms under the OASIS IPR Policy
Additional Non-normative information regarding the start-up of the TC:
a. Identification of similar or applicable work:
The proposers are unaware of any similar work being carried on in this exact
area. However, this TC intends to leverage the products of, and seek liaison
with, a number of other existing projects that may interoperate with or provide
functionality to the EKMI TC's planned outputs, including:
OASIS Web Services Security TC
OASIS Web Services Trust TC
W3C XMLSignature and XMLEncryption protocols and working group
OASIS Digital Signature Services TC
OASIS Public Key Infrastructure TC
OASIS XACML TC (and other methods for providing granular access-control
permissions that may be consumed or enforced by symmetic key management)
b. Anticipated contributions:
StrongAuth, Inc. anticipates providing a draft proposal for the EKMI protocol,
at the inception of the TC. The current draft can be viewed at:
http://www.strongkey.org/resources/documentation/misc/skcl-sks-protocol.html
and a working implementation of this protocol is available at:
http://sourceforge.net/projects/strongkey for interested parties.
c. Proposed working title and acronym for specification:
Symmetric Key Services Markup Language (SKSML), subject to TC's approval or
change.
d. Date, time, and location of the first meeting:
First meeting will be by teleconference at:
Date: January 16, 2007
Time: 10 AM PST, 1PM EST
Call in details: to be posted to TC list
StrongAuth has agreed to host this meeting.
e. Projected meeting schedule:
Subject to TC's approval, we anticipate monthly telephone meetings for the first
year. First version of the protocol to be voted on by Summer 2007. StrongAuth is
willing to assist by arranging for the teleconferences; we anticipate using
readily available free teleconference services.
f. Names, electronic mail addresses, of supporters:
Ken Adler, ken(AT)adler.net
June Leung,June.Leung(AT)FundServ.com
John Messing, jmessing(AT)law-on-line.com
Arshad Noor, arshad.noor(AT)strongauth.com
Davi Ottenheimer, davi(AT)poetry.org
Ann Terwilliger, aterwil(AT)isa.com
g. TC Convener:
Arshad Noor, arshad.noor(AT)strongauth.com

Hi Bilge,
did you put your text in a blender before sending it?
I understood everything works fine except the miscellaneous menu item in the configuration tab of ERM?
Have you already tried to clear all browser cache, close all browsers and try it again?
Best,
Frank

CUCM: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls

Hi Team,
we are running CUCM 9.1(2a),
we have integrated Third Party SIP Phone(Avaya 1230 SIP Phone) with CUCM,
Issue: Third Party SIP Phone "Caller ID" is not displaying for outgoing calls, we are able to see only the dailed Number,
When "A" calls to "B", "A" can see only the dailed number of "B" but not the "Caller ID"
Regards
Ananthakumar

Are A and B both Avaya phones?
So it looks like you're not seeing the alerting name/connected name getting updated then? Do you have alerting names configured on the directory numbers? Might need to take a look at the SIP messaging to see if the alerting name/connected name is being sent to the Avaya phones and maybe they just aren't displaying it. Might just be something that needs to be tweaked in the 46xxsettings.txt file.

New issue , when importing new paid app spreadsheet into Configurator iTunes attempts to download an ipa file for each redeem code . If you have 100 codes 100 downloads will start. Call enterprise support if you are seeing this issue.

New issue , when importing new paid app spreadsheet into Configurator iTunes attempts to download an ipa file for each redeem code . If you have 100 codes 100 downloads will start. Call enterprise support if you are seeing this issue.

New issue , when importing new paid app spreadsheet into Configurator iTunes attempts to download an ipa file for each redeem code . If you have 100 codes 100 downloads will start. Call enterprise support if you are seeing this issue.

How to get logs from CUCM for initiated calls

Dears,
like in Voice Gateway for initiated calls we can run debugs command like debug voip ccapi inout to know the calling and the called number and the status of the call, matched dial-peer, ... etc
Can we do the same in CUCM
Thanks in advance for your help

CDR is used mainly for call statistics, it is not designed for detailed troubleshooting. Reading cucm logs requires a detailed understanding of how different components function and some practice. It can be a little daunting.
This book is excellent. This is what I always recommend as the starting point..
http://books.google.co.in/books?id=YcCjKsssklIC&pg=PA42&lpg=PA42&dq=call+manager+trace+reading&source=bl&ots=ZW5_hvNFGA&sig=22rtALJZjQFfe9JrHjgAB_GgtRU&hl=en&ei=4qT3TLfTDsuYOs_I3YwI&sa=X&oi=book_result&ct=result&redir_esc=y#v=onepage&q=call%20manager%20trace%20reading&f=false
This doc explains how to read h323 traces in cucm
https://supportforums.cisco.com/docs/DOC-11779
This blog explains how to understand sip traces..
https://supportforums.cisco.com/document/113271/understanding-sip-traces

CUCM sending e-mails for missed calls..

We are waiting for a solution to get missed calls to exchange/outlook - like other vendors ( Siemens or MS-OCS with the ESTOS GW ) who can deliver such urgent needed features. Or is there a solution from Cisco already in place ?

Hi Patrick,
thanks for your response.
We have already OCS with RCC in place. But its like with CUPC. We get only information for missed calls on Computers when the client is running. This is true for MOC and for CUPC. The problem is for users who are on tour and want to look into a computer system for missed calls. This can be Outlook or another system which they can access from remote. This could be the also the webinterface of the Ciscop IP Phone. But there is no authentication nor Call history implemented.

User@IP-of-cucm showing up in call history

I have several different types of telepresence codecs running TC 7.2 registered to CUCM 10.5. I have a SIP trunk to a VCS control. If I make an outbound call from CUCM to the VCS domain, it shows up in the call history of the cucm endpoint as user@ip-of-cucm, rather than the display name configured in the endpoint I am calling (please note this is not a "Use Fully Qualified Domain Name in SIP Requests" issue. I have this option checked and the far end does see the caller ID as a SIP URI with FQDN). If I call from that same VCS registered endpoint into the CUCM registered endpoint, then the correct display name shows up in the call history of the CUCM endpoint. I've combed through a lot of SIP settings on the CUCM side but haven't found the right one yet. Does anyone know what settings I should be changing?
Thanks, Mike

Hi Mike,
We are facing the same issue for a call between DX70 and SX20. The SX 20 is running TC 7.1
If i upgrade the SX 20 to TC7.3, would the issue be resolved or will it still persist as the domain names of our CUCM and VCS are different ?? Given below is the call flow
DX70 --> SIP Route Pattern --> VCS SIP Trunk --> SX20

Cisco Jabber for Telepresence 4.6.3 Setting for "Webex Telepresence" Cloud Service

I am looking for the server setting that can be installed into the Cisco Jabber for Telepresence Client (MOVI) that will allow it to connect to the "Webex Telepresence" cloud service. I cannot find a download anywhere that has the infomation re-configured for the "Webex Telepresence" service.
Thanks for your help in advance
Tim
Well after a few hours of searching i have the Answer
The Client can be downloaded from: http://download.telepresence.webex.com/MCX/4.6.3.17194/WIN/JabberVideo_4.6.3GA.exe
This will come with the following setting pre-configured: Sign-in Setting > Internal Server: https://boot.telepresence.webex.com/tmsagent/api/rest/devices/movi/provisioning
If you have the JabberVideo Client installed and just want to change where it gets provisioned from VCS or Webex Telepresence then you should be able to use your current client and just change the internal server settings and clear the external setting.
Then you will need to use your Webex Telepresence Login ID and Password.
Hope this helps Someone.
Tim
Message was edited by: Timothy Shire

A1: Because it can use either (webex or CUPS), there are two deployment modes, all is on the documentation if you haven't gone thru it.
You type in what it's asking, nothing more. Choose one, and then use the CUPS hostname, DNS is basically mandatory, the PC needs name resolution for all you have configured (VM/CUCM/etc)
A2: No, it can be IM&P, full UC and phone mode. No, Full UC and phone mode use the PC for media termination, deskphone control uses the phone for media. Did you change the parameters during install for phone mode??? If not, it's not in phone mode.
A3: No, that can be many things, DNS, connectivity, credentials, etc
You mention DNS is not used, so, most likely that's the problem.
I'd strongly recommend you to review the whole deployment guide for Jabber for Windows.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

Unable to select 'Use Phone for audio calls' on Cisco Jabber Client 9.6 for Windows

Hi,
I have recently deployed Cisco Presence Server and integrated with Call Manager 9.1.2. I have successfully deplyed 6700 users on IM & Presence. Some of the users requested for Cisco Jabber with phone control.
I have added CFS client on the Call Manager and associated it with the same extension numbers from their desk phones. I am currently able to make audio and video calls for these specific users. I am currently using Cisco Jabber Client 9.6 for windows. I have users both daisy chained to their desk phones and who are not. Can you please confirm if it does make any difference.
Problem Faced -
I am not able to use to option use phone for audio calls. The phone comes down with a cross sign on it. At the same time Cisco Jabber by default uses the client and it works as expected.
Can you please let me know if any of you guys have faced a similar issue.
Please let me know if you need any information regading the configuration used.
Looking forward for your valuable comments.
Thank you.
Regards,
Joseph Chirayath.

Hi Will,
I am attaching the screen shot for the END USER on CUCM 9.X that has been configured on Cisco Jabber.
Please do let me know if you need any further information.
Thank you.

CUCM 8.6 Dropped call transfers involving SIP phones

Hi All,
I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone. This scenario works:
Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
These scenarios do not work:
SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
SIP phone calls Cisco phone, which transfers the original call to another SIP phone
I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem. I am just unable to understand what the trace means:
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
G711U
G711A
G722
ILBC
GSM
and our SIP software is also set to accept the first codec offered by the remote side. It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492). The trace snippet shown above is from this log.
I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it. I can try to provide more info about our CUCM configuration if needed.
Thanks in advance!

Leslie, so here is what I found from the traces....
To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
SCCP
When the transfer key is pressed
1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
2.CUCM tells the held party to listen MOH from MOH server
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
For SIP signalling. when the first transfer key is pressed
1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
lets look at succesful SCCP-----SCCP-----SIP without MTP
Point 4 above
++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
(0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
Point 5 above
++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
(0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
Point 6 Above
+++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
(0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
(0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
+++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885626,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0
m=audio 24560 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=inactive-----------------------------------------------------Inactive
Still part of Point 6 for SIP signalling
++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885628,NET]
SIP/2.0 200 OK
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
a=ptime:20
a=recvonly-------------------------------------a=recvonly
Finally Point 7 above..
+++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885630,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
+++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885634,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
Contact:
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
Call-ID: [email protected]
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
m=audio 16574 RTP/AVP 9 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
+Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
(0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
(0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
+++++++++++=Next Phone sends its ACK+++++++++++++++
(0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
+++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885635,NET]
ACK sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
Date: Tue, 19 Feb 2013 21:44:45 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.137
b=TIAS:64000
b=AS:64
t=0 0
m=audio 20352 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Now at this point all is well...and the call is connected....
Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881160,NET]
ACK sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:38:50 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Max-Forwards: 70
CSeq: 102 ACK
o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
t=0 0
m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendonly---------------------------------------------------------sendonly
+++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881161,NET]
INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:39:04 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 164
v=0
o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
t=0 0
m=audio 4000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=inactive---------------------------------------------------------------------media inactive
At this point, we should get a response back from the sip phone...
and here is what we got..
++Trying which is expected++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
[881162,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 103 INVITE
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Content-Length: 0
++++++++Then we get a BYE+++++++++++++++
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
[881163,NET]
BYE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
Contact:
Max-Forwards: 70
From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.8
To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 3 BYE
Content-Length: 0
So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
SIP------Media------MTP------------Media-------SCCP Phone
When the new destination is dialled and transfer is commited,
SIP-------------media----MTP--------media---------MTP
The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
++++++++Ivite to 492 ++++++++++++++
INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 214
v=0
o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
t=0 0
m=audio 25038 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++Invite to 491 +++++++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
[885429,NET]
INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195----------------------------------------MTP
b=TIAS:64000
b=AS:64
t=0 0
m=audio 25030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Wao! That was a long one isnt it...It was fun too.
So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Is it possible to embed Jabber Video for Telepresence in a webpage?

Hi, I wanted to ask whether it is possible to incorporate Jabber Video for Telepresence in a webpage so that (registered) users can join or make a call on a website? Or is it only possible to download the application to a PC and run it from there? Thanks!

That's correct - it is in no way possible to incorporate Jabber Video for Telepresence directly into a web browser, the client must be downloaded and installed on the PC/Mac and/or iPad.
See your other thread regarding multiple participants;
https://supportforums.cisco.com/thread/2211227?tstart=0
/jens
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FORM get_filename CHANGING p_filename.
DATA      : lv_filename TYPE string,
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