Optional RTPs

Hi All ,
Can we craete Optional Run time prompts ? I have to craete two RTPs but the value can be given to any one of these RTPs . But when I created RTPs , it seems thay tagged as Mandatory and forcing users to enter the value in every RTP. Is there any option please ?

I am not sure what you meant by optional RTP. Your script would be referencing both the RTPS. If one is optional, what would the script looks like if a value is not given in Planning?
(Your script can have more than one RTP. When you launch them in Planning, the script is scanned for all the RTPs and are prompted for values. Once the values are given (assuming nothing is set to hidden), then the script is regenerated by replacing all the refrences of RTPs with their value.). So as Celcin said, what is the use case here?
Calc Team

Similar Messages

Run Time Prompts on a Planning form - EMPA mode

Hello,
I have a Planning form in EPMA mode setup to capture latest selected members via drop down boxes in the POV section and have the business rule execute base on the latest selected members in the POV section.
This is what I have done so far ..
1) I defined the PROPERTY for each variable as "member".
2) I selected the dimension name under the DIMENSION box.
3) I defined the limit for members selection under the LIMIT box.
4) I defined the default member under DEFAULT VALUE box.
5) I enabled option "Use Last Entered Value".
6) I enabled option RTP and entered the RTP text.
With the above setup, I thought when I clicked the save button the business rule would run based on the latest selected members via drop down boxes in the POV section. However, the rule kept using the default members that I had defined in the DEFAULT VALUE box.
Am I missing something else here? Or is this another software bug?
Thank you so much for your help.

Just to drill down ur question...
have to tried to run business rule from Tools->Business Rule ?
Is it still taking the defualt values?

Capture and RTP on a SIGNED applet.

I am trying to build an applet that allows users to chat both ways. I want the user to install nothing (other than java) and i provide the libraries they need (including jmf). I have signed the applet to ease my troubles, but it seems they are still there. The basic goal: capture audio from two microphones, send the audio via RTP, hear audio on both sides.
Even though I have a signed applet with all permissions, I lack permission to capture from applets? Aside from the fact that this seems absurd, it's frustrating. I am looking to find a way around this. I don't want the users to have to manually change the JMFRegistry or download a program to it.
The better options :
1) Is there a way to take advantage of the signed applet to alter the permissions in the program? (Seems very unlikely, but it would be the best).
2) Is there a way to provide the permissions in the jmf.jar (including jmf.properties?) that I include? (Seems more unlikely than 1)
3) What makes this more interesting (frustrating), I can use the Java Sound API to access the microphone and open an AudioInputStream. Can I plug this stream, or some other object from javasound, into the JMF somehow to send it via RTP?
4) Other options?

I found a way to use javasound instead of the jmf for the source audio (that's number 3 in my original post). So, i'm good for my problem.
If you want to do mixing, javasound should have more support for mixing (though I've not done it). Check out this thread, it was very useful for me.
http://forum.java.sun.com/thread.jspa?threadID=680525&start=0&tstart=0
I might have spoken too soon... I am just getting static - but that might be related to audio formats.....
Message was edited by:
mortserg

RTP in Calc manager

Hi,
I have used an RTP variable in CM rule and used that rule in multiple rule sets.
I want to run the rule sets with different default value of RTP in those rule sets.
This was possible in HBR sequence but I don't know how can I achieve here in CM.
I have to use the same rule in 10 different rulesets and I don't wnat to repeat the same rules
with different coded values.
let me know how can we do this...
Thanks,
CM

HI,
You can design CM rulesets w/out using of option "Merge Variables". So you can set different values for RTP variables.
http://docs.oracle.com/cd/E17236_01/epm.1112/cmgr_admin/cmgr_rulesetdesigner.html

H.263/RTP payload

As we know jmf can stream in h.263/rtp with payload 34. i am streaming h.263/rtp to a device which does not understand payload 34 rather payload 96 whihc is used by h.263-1998 and 2000 updated versions.
Do anyone has any idea how can change payload from 34 to compatible with 96.
Sorry if it is not the right place/forum to ask this question. But if any one has any idea where i can ask question regarding transcoding h.263/rtp packets from the one compatible with rfc 2190 (which is supported by jmf and is almost outdated) to packets compatible with rfc 2429.

Sorry captfoss i was on leave for some time so could not respond to your reply on time.
While working today i found that JMF has a videoformat :-VideoFormat.H263_1998_RTP which is compaitable with RFC 2429. But there is a problem - i tried several times to transcode my h263 file i am reading from local computer to h263_1998_RTP format but it fails giving following error:-
Failed to realize: com.sun.media.ProcessEngine@19209ea
Cannot build a flow graph with the customized options:
    Unable to transcode format: H263, 176x144, FrameRate=15.0, Length=76032 0 extra bytes
      to: H263-1998/RTP
      outputting to: RAW/RTP
Error: Unable to realize com.sun.media.ProcessEngine@19209eathen i checked in jmf registery i found a codec -com.sun.media.codec.video.vh263.Native Decoder -This Decoder converts h263,h263/rtp as well as h263-1998/rtp into yuv format. but i could not found any encoder which converts any format(yuv or rgb or h263 or any other) to h263-1998/rtp.
Is there any way by which i can transcode to h263-1998/rtp???

High Delay of MPEG-1 P-Frames first slices in RTP Streams

Hi,
I'm trying to implement a simple RTP server using JMF 2.1.1e (Windows Performance Pack). I started off with the sample application available at
http://java.sun.com/products/java-media/jmf/2.1.1/solutions/ToolsTx.html
For receiving and displaying the MPEG-1 stream I'm using JMStudio. Basically, the transmission seems to work and JMStudio displays the video I'm transmitting from the server computer.
Unfortunately, the video does not display fluently / smoothly, the frame rate fluctuates between 10 and 30 fps for a video stream that was originally 25 fps.
Trying to track down the source of the problem, I used Etherreal to sniff the network traffic. According to this, the server seems to have problems espacially with the first slice / packet of most P-Frames. They are delayed for more than 0.1s (normal delay seems to be round about 0.0004s).
Using this insight, I tried MPEG-1 videos using other GOP structures.
1. IPPPPPPPPPPPPPP: As one would expect, using this GOP, the video displays even worse, since almost all frames are P-Frames.
2. I: Using only I-Frames, the problem almost disappears and the video displays very smoothly.
This leads me to the conclusion that inside JMF there seems to be a problem while packetizing the P-Frames.
Did anybody experience similar problems or maybe even has a solution for this problem?
Any help would be greatly appreciated!
Kind regards
Sebastian Seitz

Dag Norum:
First of all, thank you for your quicly reply ;-)
Second, I'm very sorry to make a reply almost a month after your answer. The reason is that I went on holidays without an opportunity to test your tip.
I used Harm's link and than reached to a Bitrate of 8.0000 for all three fields (Min., Target and Max.) and that makes all the difference. Now I have a video with very good quality.
But I haven't tested your tip yet to see if that improves the video final quality.
I want to work always with the maximum quality, no matter what time is needed to convert the video and no matter the length (GB) it costs.
Do you think I can export to an uncompressed file (avi for windows and compressor set to none) and than make the Menus with Encore? It wouldn't be better work on Encore with an MPEG2 file? The DV compressor is MPEG2, isn´t it?
And why you suggested to not have the "Optimize stills" checked? This is to optimize the frames without movement, isn´t it? Many times I use photos inserted in the movie. This option isn't good for that?
Your last suggestion, to go directly from timeline to the end target (MPEG2) is much different than that I use (DV compressor)?
If you could answer my questions I would appreciate.
Many thanks!
(sorry if my english isn't the better)
Message was edited by: Warlord_LA (01-Oct-2009 23h51)
Sorry, I'd made a mistake. When I answer to you I didn't realise that your sugestions was to AVI exportation and not to MPEG2.
Of course you are absolutely right: for an intermediate file, no compression is always better than some compression :-)
But my question about not have the "Optimize stills" checked remains. What does it makes, really?

Communication between Lync - Mediant . 488 evt from mediant site when SBC Remote Early Media RTP parameter is set to Delayed

Hi,
I have a problem with Early Media functionality between Lync and Mediant 1000 E-SBC. If SBC Remote Early Media RTP parameter is set to Delayed (on IP Profile Settings level) i cannot establish connection. After receiving 183 Session progress (from Lync site)
call is released with 488 event (488 is sent from mediant site). Before "Not applicable here" in logs i can see:
6d:9h:46m:35s (      lgr_flow)(18219     ) #MediaResourcesConnector::AllocateMediaResources
(lgr_media_connector)(18220     ) MediaResourcesConnector:CalculateResourcesForAppExtensions Leading:DSP Opposite:NONE MediationLevel:NONE
(lgr_media_service)(18221     ) ServicesMngr: Allocate Media channel. current active: 0 and max is: 0
6d:9h:46m:35s (lgr_media_service)(18222     ) !! [ERROR] ServicesMngr: Cannot allocate more Media channel. current active: 0 and max is: 0
6d:9h:46m:35s (lgr_media_service)(18261     ) (#292) ChannelResource Deallocated.
6d:9h:46m:35s (lgr_media_service)(18262     ) (#293) ChannelResource Deallocated.
I need this because if im using SBC Remote Early Media RTP = Immidiately i cannot hear ringtone. Could someone give me any clues what is wrong in config?
I enclosed detailed logs:
Log is Activated 6d:9h:46m:35s (       lgr_sbc)(18057     ) (#51) SBCRoutesIterator Allocated.
(       lgr_sbc)(18058     ) Classification Succeeded - Source IP Group #2.
(       lgr_sbc)(18059     ) CallAdmission::AddDialog: Type INVITE Direction In IP group 2: 1 SRD 2: 1 SRD ovflw:1 used unres:1
(      lgr_flow)(18060     ) (#51)SBCRoutesIterator::Change State From: InitialRouting To : AlternativeRouting
(       lgr_sbc)(18061     ) CallAdmission::AddDialog: Type INVITE Direction Out IP group 1: 1 SRD 1: 1 SRD ovflw:1 used unres:2
(      lgr_flow)(18062     ) (#51)SBCRoutesIterator::Next route found: Route by: IPGroup , IP Group ID: 1, Live:1
(       lgr_sbc)(18063     ) Routing Succeeded -IP2IPRouting Rule #2.
(      lgr_flow)(18064     ) ---- Incoming SIP Message from switch:5060 to SIPInterface #2 UdpTransportObject[#2] ----
6d:9h:46m:35s INVITE sip:1234@Mediant;user=phone SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
Supported: replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R10.1.1 j2.603.33.a
Session-Expires: 1800;refresher=uac
Min-SE: 900
P-Asserted-Identity: "TPSA 12" <sip:221111111@switch;user=phone>
Content-Type: application/sdp
To: <sip:1234@Mediant;user=phone>
From: "TPSA 12" <sip:221111111@switch;user=phone>;tag=434636a077aeb3418ef3d5c0887debbd
Contact: <sip:221111111@switch;transport=UDP>
Call-ID: 5050e7eb34568795c049d14a6b088172@switch
CSeq: 1290334631 INVITE
Via: SIP/2.0/UDP switch;branch=z9hG4bK668ced008893ef177292c6f0f662ca02
Max-Forwards: 70
Content-Length: 265
v=0
o=OXE 1415263596 1415263596 IN IP4 switch
s=abs
c=IN IP4 172.16.20.32
t=0 0
m=audio 32156 RTP/AVP 8 4 97
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:4 G723/8000
a=ptime:30
a=maxptime:30
a=rtpmap:97 telephone-event/8000
(     sip_stack)(18066     ) New SIPMessage created - #57
(     sip_stack)(18067     ) New SIPSBCCallLeg created - #1695
6d:9h:46m:35s (     sip_stack)(18068     ) New AcSIPCall created - #1922
(     sip_stack)(18069     ) AcTransactionUser::AddMessageToQueue: Queueing message
6d:9h:46m:35s (     sip_stack)(18102     ) SIPCall(#1922) changes state from Idle to Invited
6d:9h:46m:35s (     sip_stack)(18164     ) SIPCall(#1921) changes state from Idle to Inviting
(     sip_stack)(18165     ) SIPSessionTimer<TU#1921>::FillSTRequestData - Session-Timer mode: TRANSPARENT
(      lgr_flow)(18166     ) ---- Outgoing SIP Message to Lync2:5068 from SIPInterface #1 TcpTransportObject[#648] ----
6d:9h:46m:35s (      lgr_flow)(18178     ) ---- Incoming SIP Message from Lync2:5068 to SIPInterface #1 TcpTransportObject[#648] ----
SIP/2.0 183 Session Progress
FROM: "TPSA 12"<sip:221111111@switch;user=phone>;tag=1c1656521459
TO: <sip:1234@Mediant;user=phone>;tag=499c98c8cd;epid=DE8F26F104
CSEQ: 1 INVITE
CALL-ID: 1656462877611201494635@Mediant
VIA: SIP/2.0/TCP Mediant:5068;branch=z9hG4bKac1657277504;alias
CONTACT: <sip:lyfe02.grupa.lukas:5068;transport=Tcp;maddr=Lync2>
CONTENT-LENGTH: 281
CONTENT-TYPE: application/sdp
ALLOW: CANCEL
ALLOW: BYE
ALLOW: UPDATE
ALLOW: PRACK
REQUIRE: 100rel
SERVER: RTCC/5.0.0.0 MediationServer
Rseq: 1
v=0
o=- 89 1 IN IP4 Lync2
s=session
c=IN IP4 Lync2
b=CT:1000
t=0 0
m=audio 52882 RTP/AVP 8 97
c=IN IP4 Lync2
a=rtcp:52883
a=label:Audio
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
m=audio 0 RTP/SAVP 8 4 97
(     sip_stack)(18180     ) New SIPMessage created - #55
(      lgr_flow)(18181     ) |       |(SIPTU#1921)183 State:Proceeding(1656462877611201494635@Mediant)
6d:9h:46m:35s (   lgr_stk_ses)(18182     ) |       |       |       #1694:SIP_ALERT_EV(1656462877611201494635@Mediant)
(     sip_stack)(18183     ) SDPBody::New - id = 126
(     sip_stack)(18184     ) SDPBodyMedia::New - id = 541
(     sip_stack)(18185     ) SDPBodyMedia::New - id = 540
(     sip_stack)(18186     ) <BaseSIP SDPSESSION #1694> UpdateChosenMediaByCN - CN as Remote -1
(     sip_stack)(18187     ) SDPBodyMedia::New - id = 539
(     sip_stack)(18188     ) SDPBodyMedia::New - id = 538
(   lgr_stk_ses)(18189     ) SBCOfferAnswerMngr(#1694) changes state from SIP_MEDIA_OFFERING to SIP_MEDIA_COMPLETED
(   lgr_stk_ses)(18190     ) <SESSION #1694> SendToCall - event: MEDIA_NEGOTIATION_COMPLETED_EV m_Call#1852
(      lgr_flow)(18191     ) |       |       (#1852)SBCCall <- (#1694)SIPSBCCallLeg: MEDIA_NEGOTIATION_COMPLETED_EV
(      lgr_flow)(18192     ) |       (#59)SBCCallPlacementFeature <- (#1852)SBCCall: MEDIA_NEGOTIATION_COMPLETED_EV
(      lgr_flow)(18193     ) |       (#1853)SBCParticipantEndPoint <- (#59)SBCCallPlacementFeature: MEDIA_NEGOTIATION_COMPLETED_EV
6d:9h:46m:35s (      lgr_flow)(18219     ) #MediaResourcesConnector::AllocateMediaResources
(lgr_media_connector)(18220     ) MediaResourcesConnector:CalculateResourcesForAppExtensions Leading:DSP Opposite:NONE MediationLevel:NONE
(lgr_media_service)(18221     ) ServicesMngr: Allocate Media channel. current active: 0 and max is: 0
6d:9h:46m:35s (lgr_media_service)(18222     ) !! [ERROR] ServicesMngr: Cannot allocate more Media channel. current active: 0 and max is: 0
6d:9h:46m:35s (lgr_media_service)(18261     ) (#292) ChannelResource Deallocated.
6d:9h:46m:35s (lgr_media_service)(18262     ) (#293) ChannelResource Deallocated.
6d:9h:46m:35s (      lgr_flow)(18263     ) |       |       (#1853)SBCCall <- (#1695)SIPSBCCallLeg: RELEASE_EV
(      lgr_flow)(18264     ) |       |       (#1853) SBCCall changing states from:AlertingState to:DisconnectingState
(      lgr_flow)(18265     ) |       (#1852)SBCParticipantEndPoint <- (#1853)SBCCall: RELEASE_EV
(      lgr_flow)(18266     ) (#1852) SBCParticipantEndPoint changing states from:InitiatedState to:ReleaseingState
(      lgr_flow)(18267     ) (#1035)SBCController <- (#1852)SBCParticipantEndPoint: RELEASE_EV
(      lgr_flow)(18268     ) |       |       (#1035) SBCController changing states from:EstablishingState to:DisconnectingState
(      lgr_flow)(18269     ) |       (#1853)SBCParticipantEndPoint <- (#1035)SBCController: RELEASE_EV
(      lgr_flow)(18270     ) (#1853) SBCParticipantEndPoint changing states from:InitiatedState to:ReleaseingState
(      lgr_flow)(18271     ) |       (#59)SBCCallPlacementFeature <- (#1853)SBCParticipantEndPoint: RELEASE_EV
(      lgr_flow)(18272     ) (#59) SBCCallPlacementFeature changing states from: Initiated to: Releasing
6d:9h:46m:35s (      lgr_flow)(18273     ) |       |       (#1852)SBCCall <- (#59)SBCCallPlacementFeature: RELEASE_EV
(      lgr_flow)(18274     ) |       |       (#1852) SBCCall changing states from:AlertingState to:DisconnectingState
(      lgr_flow)(18275     ) |       |       |       (#1694)SIPSBCCallLeg <- (#1852)SBCCall: RELEASE_EV
(     sip_stack)(18276     ) New SIPMessage created - #52
(      lgr_flow)(18277     ) |       |(SIPTU#1921)DISCONNECT_REQ State:Proceeding(1656462877611201494635@Mediant)
(     sip_stack)(18278     ) New SIPMessage created - #49
(      lgr_flow)(18279     ) ---- Outgoing SIP Message to Lync2:5068 from SIPInterface #1 TcpTransportObject[#648] ----
CANCEL sip:1234@Mediant;user=phone SIP/2.0
Via: SIP/2.0/TCP Mediant:5068;branch=z9hG4bKac1657277504;alias
Max-Forwards: 70
From: "TPSA 12" <sip:221111111@switch;user=phone>;tag=1c1656521459
To: <sip:1234@Mediant;user=phone>;epid=DE8F26F104
Call-ID: 1656462877611201494635@Mediant
CSeq: 1 CANCEL
User-Agent: E-SBC/v.6.60A.260.002
Reason: SIP ;cause=488 ;text="488 Not Acceptable Here"
Content-Length: 0
(     sip_stack)(18281     ) Resource SIPMessage deleted - #49
6d:9h:46m:35s (     sip_stack)(18354     ) New SIPMessage created - #42
(     sip_stack)(18355     ) Resource SIPMessage deleted - #42
(      lgr_flow)(18356     ) |       | TransactionUserMngr::ReturnTransactionUser - #1921
(     sip_stack)(18357     ) SIPCall(#1921) changes state from Disconnected to Idle
6d:9h:46m:40s (     sip_stack)(18358     ) New SIPMessage created - #47
(     sip_stack)(18359     ) Resource SIPMessage deleted - #47
(      lgr_flow)(18360     ) |       | TransactionUserMngr::ReturnTransactionUser - #1922
(     sip_stack)(18361     ) SIPCall(#1922) changes state from Disconnected to Idle

Hi,
On Mediant 1000 E-SBC SBC tab, when set Remote Early Media RTP Behavior to
Delayed, but also check if the following configuration set correctly:
SBC Media Security Behavior: SRTP
PRACK Mode: Optional
Remote Update Support: Supported Only After Connect
Remote Re-INVITE: Supported Only With SDP
Remote Delayed Offer Support: Not Supported
Remote REFER Behavior: Handle Locally
Remote 3xx Behavior: Handle Locally
Enforce MKI Size: Enforce
You can refer this
link
Note: Microsoft is providing this information as a convenience to you. The sites are not controlled by Microsoft. Microsoft cannot make any representations regarding the quality, safety, or suitability of any software or information found there.
Please make sure that you completely understand the risk before retrieving any suggestions from the above link.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Need to restrict some hierarchy in RTP member selection

Hi There,
We are using on rule which conatins RTP variable.As per our requirement while selecting that RTP varaibale member through member selection window can I restrict the dimension hierarchy to only level 0 members as there are many levels in this dimension and we only want level zero members list to select from them.
I have tried with @levmbrs(dim,0) in Limits section but without success.
Thanks

Thanks Alp for your prompt reply.
I tried this option but nothing happened.Do you think by using the limit in variables(RTP) the pop up window while launching the rules will contain only the level 0 members as I described in the limit section ?
For example: The dimension hiearrchical structure is as below and I need to show only Entity11,Entity12,Entity21 in the member selection window .Currently it is showing the below structure in the pop up while clicking the member selection icon.
Entity
-->Entity1
------>Entity11
------>Entity12
-->Entity2
------>Entity21

PA: module-rtp-recv: Sink does not exist

Hello Archers,
I really would like to route my primary computer's sound via RTP to a remote machine (a laptop, which is used as my home server).
Both machines have Pulseaudio set up and local playback on both works fine.
Unfortunately on the remote machine, I cannot load the RTP receive sink. I tried the default PA package (1.1-1) and compiled pulseaudio-git, but still the same issue.
$ pulseaudio --version
pulseaudio 1.98-dev-186-g07e47
$ pulseaudio -nC
Welcome to PulseAudio! Use "help" for usage information.
>>> load-module module-rtp-recv
>>> E: [pulseaudio] module-rtp-recv.c: Sink does not exist.
$ ls -la /usr/lib/pulse-1.98/modules/ | grep rtp
-rwxr-xr-x 1 root root 38460 Dec 28 13:38 librtp.so
-rwxr-xr-x 1 root root 26264 Dec 28 13:39 module-rtp-recv.so
-rwxr-xr-x 1 root root 17984 Dec 28 13:38 module-rtp-send.so
Configs on remote machine:
default.pa
system.pa
The only thing I could find on Google was the same issue with a 0.9 version, which seemed to be a bug in PA. But I cannot imaging, this is still an issue, as I think some of you use an RTP setup with PA successfully.
Is there any possibility to debug that issue?
Thanks in advance,
R.

It seems to be internal to Pulse Audio, which is what I use by default. I don't even think I use gstreamer.
It works fine if I change it manually in pavucontrol, I just want the option to stick. Any other ideas?

SIP Options Ping

Cisco CME to Cisco UBE Options ping no-worky....
Did anyone get SIP options pings working on a pure Cisco router (UBE to UBE and UBE to CME) environment? I have a pure Cisco 15.1 network that we are playing with option pings on. I see pings go out and 200 OK responses, but dial-peers are still busying out. Have tried both default and example settings from Cisco UBE deployment guides. Routers are working with options pings from external ITSPs, so know options replies are working and are supported. So, what's the magic sauce on Cisco IOS 15.1 for SIP options-keepalive?
dial-peer voice 40221 voip
corlist outgoing peer-internal
description \\\ 4385 - 4389 \\\
translation-profile outgoing ToCUCME_NANP_PrefixedE164
preference 1
destination-pattern 438[5-9]
session protocol sipv2
session target ipv4:10.5.5.1
voice-class sip early-offer forced
voice-class sip options-keepalive up-interval 12 down-interval 65 retry 3
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-throug
no vad

There are no firewalls or NAT between and when using "debug ccsip message" we see option sent and received and replies on both sides.

SAP ISU RTP Price Profile

I am trying to configure RTP for real time scenarios like index prices which will be coming as input from market.We have done some configuration on this but on billing we are getting bill as Normal bill i.e. Total amount . I want to see the calculation as per the intervals ? e.g every 30 mins.
Also If I have to attach prices for every 30 mins How can I do that ? If we have to use price profile ...then we cannot create a RTP operand for price as RTP operand will be used only for Quantity,demand & Amount.
Any help will be appriciated.
thanks

I think you have 2 options:
1. If you just want to bill each 30 mins with a price then you can create a formula profile for each customer that is to be billed like this.
The profile should have PV category Amount and as a formula you may use LIMIT02. Define the limit as 0 and all your consumption will be above the limit and evaluated with the price profile.
You would then have to calculate your formula profile to be able to bill the customer. The RTP interface can then calculate the total amount and if you require eg. peak demand.
2. You can do it completely in the RTP interface.
Define a price profile and use this as an input parameter, either by role or enter the specific profile number in the RTP interface, to your formula.
As the other input parameter use your quantity profile, allocated by role.
Then your formula will calculate price * quantity and return a profile. The result functions in the RTP interface result parameters make it 1 value (the sum, minimum or maximum or any other function you define).
Make your formula store the calculated result profile as well, by finding the result profile to store on based on the input profile (allocation to register, profile role to find right profile).
Option 1 is the most simplistic and would need no custom development.

UCCX 8.5 - How do I create a menu option that overwites a prompt?

Hi All,
I'm fairly new to UCCX scripting and have a question about a way to create a menu option that overwrites a prompt.
Here is the situation, we have only one small call center. There are two scripts in play here, one is their production script. Everythign is working fine with that script. The other is a recording script. The call center supervisors would like to be able to call into the recording script and record a prompt that says something like "All agents will be out today, November 1st, until 2pm on a staff luncheon, please call back at 2pm". This would overwrite the welcome prompt on their production script. As you can see from the language above, the message would be different depending on what was going on that day. I actually have this working quite well.
The problem is that when they come back from their luncheon, or from whatever they were doing, they want the ability to revert back to the original opening greeting wav file. The way it's set up now, when they get back from lunch they'd have to re-record the whole opening greeting "Thank you for calling XXX, press 1 to be connected to an agent, press 2...".
It seems like there should be a way, a new menu option in the recording script, that would allow me to revert back to the original recording so they don't need to re-record the opening greeting every time. I've created a new folder in Prompt Managment called "Revert" and placed the orginal opening greeting there. Can anyone give me guidance on what step to insert that would in effect say, if the caller presses 3, copy "Revert/Welcome.wav" to "CallCenter/Welcome.wav"? In this case the "CallCenter" folder contains the production prompts for the CallCenter script.
Thanks,
Josh

Josh,
     I would recommend the following configuration:
          1.) Setup a XML file that has an option called "EarlyClose" or something similar.
          2.) Setup your script with an "IF" that reads in that XML document and looks for "EarlyClose". If it's set, then play the custom closed message. If not, then play the normal message that is never overwritten.
     The second part is to create a second script that has an internal only extension. This would then allow your supervisor to dial in, and put in a PIN. Once this is done, set have the agent record the greeting. Then (and only then) set the "EarlyClose" variable in your XML document. This will prevent the user not getting the new prompt recorded and EarlyClose being set early. You can also setup the script to play back the prompt and then have the end user validate that everything is working right. I would recommend reading through the scripting guides as they have some great infomation on this type of thing and example scripts. I hope this helps you out.
Thank you,
Robert W. Rogier -- UCCE TAC Support Engineer
Cisco, RTP NC.

RTP formats

Hi,
I'm using AVTransmit3 code to transmit a .wav file. I have jmf.jar (cross-platform) and at first it only found and worked with 5 formats, like these:
dvi/rtp, 8000.0 Hz, 4-bit, Mono
dvi/rtp, 11025.0 Hz, 4-bit, Mono
dvi/rtp, 22050.0 Hz, 4-bit, Mono
gsm/rtp, 8000.0 Hz, Mono, FrameSize=264 bits
ULAW/rtp, 8000.0 Hz, 8-bit, Mono, FrameSize=8 bits After, I installed the windows performance pack, and it found and worked with 12 formats, (last five, and seven more). The "new formats are:
{code}
g723/rtp, 8000.0 Hz, Mono, FrameSize=192 bits
mpegaudio/rtp, 48000.0 Hz, 16-bit, Stereo, BigEndian, Signed
mpegaudio/rtp, 44100.0 Hz, 16-bit, Stereo, BigEndian, Signed
mpegaudio/rtp, 22050.0 Hz, 16-bit, Stereo, BigEndian, Signed
mpegaudio/rtp, 48000.0 Hz, 16-bit, Mono, BigEndian, Signed
mpegaudio/rtp, 44100.0 Hz, 16-bit, Mono, BigEndian, Signed
mpegaudio/rtp, 22050.0 Hz, 16-bit, Mono, BigEndian, Signed
{code}
And now, I unistall windows perfomance pack, and I'm working again with cross-platform jar, and it find the 12 formats, but only works with the first five formats.
I have this problem:
{code}Track 0 is set to transmit as: mpegaudio/rtp, 48000.0 Hz, 16-bit, Stereo, BigEndian, Signed
Failed to build a graph for the given custom options.
Failed to realize: com.sun.media.ProcessEngine@1a28362
Cannot build a flow graph with the customized options:
Unable to transcode format: LINEAR, 16000.0 Hz, 8-bit, Mono, Unsigned, 16000.0 frame rate, FrameSize=8 bits
to: mpegaudio/rtp, 48000.0 Hz, 16-bit, Stereo, BigEndian, Signed
outputting to: RAW/RTP
Error: Unable to realize com.sun.media.ProcessEngine@1a28362
INFO: Couldn't realize processor{code}
As I read in : [http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/formats.html] these rtp formats are supported for crossplatform version (a-law not for encoded and transmit, but mpegaudio, yes)
I don`t know if i'm wrong, nor what may be failing. Any idea?
Thanks

Ok, Thanks. Now I understand it better.

Streaming H264 content via RTP/RTSP: VLC works, Quicktime doesn't

Hello,
We are developping a DSP based system that streams H264 content using RTP/RTSP.
We are feeding Quicktime and VLC with a similar URL: rtsp://212.143.39.228:12654/stream/device71_camera2.sdp
Both clients communicate with our server in the RTSP protocol. The following messages pass between Quicktime client and our server:
DESCRIBE rtsp://212.143.39.228:12654/stream/device71_camera2.sdp RTSP/1.0
CSeq: 1
Accept: application/sdp
Bandwidth: 512000
Accept-Language: nl-NL
User-Agent: QuickTime/7.6.7 (qtver=7.6.7;os=Windows NT 5.1Service Pack 3)
RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 435
v=0
o=- 4 16975365 IN IP4 212.143.39.228
s=MRS
t=0 0
b=RS:0
b=RR:0
a=range:npt=now-
a=isma-compliance:2,2.0,2
a=control:rtsp://stream/device71_camera2.sdp
m=video 5438 RTP/AVP 97
c=IN IP4 10.0.0.51
a=control:trackID=1
b=AS:250
a=rtpmap:97 H264/90000
a=fmtp:97 packetization-mode=1;profile-level-id=42801E;sprop-parameter-sets=Z0KAHomVQsEtA EA=,aFOAwGHI
a=mpeg4-esid:201
a=cliprect:0,0,288,352
a=framesize:97 352-288
SETUP rtsp://stream/trackID=1 RTSP/1.0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=6970-6971
x-retransmit: our-retransmit
x-dynamic-rate: 1
x-transport-options: late-tolerance=2.900000
User-Agent: QuickTime/7.6.7 (qtver=7.6.7;os=Windows NT 5.1Service Pack 3)
Accept-Language: nl-NL
RTSP/1.0 200 OK
CSeq: 2
Session: 4
Transport: RTP/AVP;unicast;clientport=6970-6971;serverport=12655-12656
PLAY rtsp://212.143.39.228:12654/stream/device71_camera2.sdp RTSP/1.0
CSeq: 3
Range: npt=0.000000-
x-prebuffer: maxtime=2.000000
Session: 4
User-Agent: QuickTime/7.6.7 (qtver=7.6.7;os=Windows NT 5.1Service Pack 3)
RTSP/1.0 200 OK
CSeq: 3
At this stage our server starts sending H264 RTP frames to port 6970 of Quicktime's host machine. No firewall or NAT exists between the client and server machines. Sometimes it is the same machine.
Quicktime client says:
'Negotiating'
'Waiting for media...'
and after 10 seconds it says 'Not Found' and disconnects from the server.
VLC displays the video stream given the same URL.
Why is Quicktime not finding/displaying the stream?

I have managed to pass this stage by adding an 'RTP-Info' header to the 'PLAY' reply. Now the player does buffering and starts counting the time forward as if it were doing a kind of playback, but all that can bee seen on the Quicktime window is the large blue Quicktime icon with dots below scrolling from side to side.
Movie inspector shows an advancing 'Current Time', 'Movie FPS: 0.00', 'Format: StreamingH.264, 352x288, Millions', a good Bit Rate value and a Quality near 100%.

RTP on Adobe Air

We are a broadcast solution developer in Korea.
I want to know if Adobe Air can play multimedia contents transmitted by RTP-E.S. packets on RTSP over RTP protocol or HTTP tunneling from streaming server.
I only can see UDP or HTTP are possible on Adobe Air on the website.
Thank you.

Unfortunately, I don't have a definitive list of AIR based applications. I'm sure you're aware, but the AIR runtime is only used when an application requires it. It's also possible, with our captive runtime option, for an application to be built that includes it's own copy of the runtime.
I believe that the Adobe suite installer (and possibly other online installers) also use AIR. Once installed, AIR might not be used again. Again, this really depends on the application.
When you mention that your company is looking to "slimming down our desktop loads", are you talking about system performance, drive space, or something else?

Optional RTPs

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