RTP simulation on j2me for voice transfer

hi,
i m developing a voip app for mobile. i have some probs.
can we have rtp impln on j2me ?
or can we simulate UDP as RTP ?
whats a best packet size for transferring realtime recorded voice ?
My prog reports around 8KB of data for 1sec recording.
can anyone put some light in this darker area of coding.
thank u.

Sebastien,
The 7960 phones have an option called PC Voice Vlan Access in the CCM Phone configuration. The default is on for this. If you don't want your phone traffic to be captured on the PC port disable this. Here is the documentation description.
PC Voice VLAN Access : Indicates whether the phone will allow a device attached to the PC port to access the Voice VLAN. Disabling Voice VLAN Access will prevent the attached PC from sending and receiving data on the Voice VLAN. It will also prevent the PC from receiving data sent and received by the phone. Set this setting to Enabled if an application is being run on the PC that requires monitoring of the phones traffic. These could include monitoring and recording applications and use of network monitoring software for analysis purposes.
This is a required field.
Default: Enabled.
Additionally as an FYI the 7970(and possibly some other newer phones) has a secondary option as well.
Span to PC Port : Indicates whether the phone will forward packets transmitted and received on the Phone Port to the PC Port. Select Enabled if an application is being run on the PC Port that requires monitoring of the IP Phone's traffic such as monitoring and recording applications (common in call center environments) or network packet capture tools used for diagnostic purposes. To use this feature PC Voice VLAN access must be enabled.
This is a required field.
Default: Disabled.
Please rate any helpful posts
Thanks
Fred

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Hi,
We have a 2Mbps LL 1:4
we are using CSICO ATA for Voice.
we are using cisco 2620 router .
Here are my questions.
1.Kindly check My config and say whether this QOS config will work for prioritising the Voice.
class-map match-all VOIP-RTP
match ip dscp ef
policy-map VOICE-QOS
class VOIP-RTP
priority 1024
interface Serial0/0
description ### STPI-GATEWAY-VASHI ###
bandwidth 2048
ip address 213.11.12.115 255.255.255.252
ip access-group 103 in
ip access-group 103 out
service-policy output VOICE-QOS
shutdown
2.How can i filter the HTTP,TELNET,SSH,RDP,FTP traffic.
Kindly help me.
Thanks
Ranga

A more scalable config (that you dont have to redo too much) might include bandwidth guarantees for other classes of traffic as well...
Also, I like to go with the qos design guide recommendation and set aside a queue for voice signalling... like the following...
i also dont "match ip dscp ef" but rather just look for rtp audio... dont always have a marking switch/phone system behind your router... sometimes its a whitebox phone system sending rtp packets, and a dumb switch... I also go with a nested policy, which shapes all to the speed of the link, then decides which traffic will follow the rules of the child policy to leave the single queue ;)
class-map match-any manage
match protocol dhcp
match protocol dns
match protocol kerberos
match protocol ldap
match protocol snmp
match protocol syslog
class-map match-any bulk
match protocol exchange
match protocol ftp
match protocol pop3
match protocol smtp
class-map match-any voicesignal
match protocol h323
match protocol rtcp
class-map match-any transactional
match protocol citrix
match protocol pcanywhere
match protocol secure-telnet
match protocol sqlnet
match protocol sqlserver
match protocol ssh
match protocol telnet
match protocol tsrvrdp
class-map match-any video
match protocol rtp video
match protocol cuseeme
match protocol netshow
match protocol rtsp
match protocol streamwork
match protocol vdolive
class-map match-any voicebearer
match protocol rtp audio
policy-map Pol-S0/0/0.1-child
class voicebearer
set dscp ef
priority percent 25
class transactional
bandwidth percent 25
class voicesignal
bandwidth percent 5
class manage
bandwidth percent 5
policy-map Pol-S0/0/0.1-parent
class class-default
shape average 1444000
service-policy Pol-0/0/0.1-child
int s0/0/0.1
service-policy output Pol-S0/0/0.1-parent
(yes not all my classes are used in my policy; they are for future use... nice to have them in there now though, as they can always be allocated some bandwidth later on, at the expense of what is carved out now...)
Tschuss,
Joe

VoFR PVC for data and PVC for voice with FRTS

Dear friends, It is my first experience configuring VoFR. My access rate is 128 Kbps and the CIR is 64 Kbps. I configured the router following some cisco examples using "vofr cisco" and FRTS into my "map-class" for voice. You can see my config in the attach file.
It works fine, the quality voice is good (2 calls)when the link is congested up to 100% through bi-directional ftp download.
However, I cannot get a speed upper to 64 Kbps. I understand that it is as FRTS works and it is necesary to QoS for voice. But I`m paying to my service provider a price for 128/64 and I never won`t get a speed greater than 64 Kbps.
How can I solve this problem without increase my CIR?. It has a additional cost.
I`m reading that some solution could be create two PVCs: one for voice and one for data. How should I configure them to get a burst traffic for data upper to 64 Kbps without affect my quality of voice?
Thank you for your suggestions.
Edwin.

The following document has some explaination on the queueing on frame relay interfaces. This might help you understand how to configure different pvcs for voice and data and utilise the given cir fully.
Understanding Queueing on Frame Relay Router Interfaces
http://www.cisco.com/en/US/tech/tk543/tk544/technologies_tech_note09186a00800a4754.shtml
VoIP over Frame Relay with Quality of Service (Fragmentation, Traffic Shaping, LLQ / IP RTP Priority)
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_configuration_example09186a0080094af9.shtml

Using Cisco 3526 IPVC for Voice Calls as well

Does anybody know the latest capabilities of the Cisco 3526 gateway when used as a Voice gateway? The Cisco SRND for "Cisco IP Video Telephony", which was published in July 2004, states that the IPVC gateways do NOT support H.245 Empty Capabilities Set or H.450. So they will give you supplementary services such as Hold, transfer, etc. Version 3.0 of the 3526 documentation lists "Empty capability set support" as a feature, although it does not state H.245 or H.450.
Basically, I want to use a 3526 as a second Voice gateway as well as a Video gateway in a Call Manager deployment. I am just concerned that I won't get all the supplementary services when using it for voice calls.
Any input would be greatly appreciated.

I know the product can support voice, my main concern was the support for ECS (Empty Capabilities Set) within H.245. If it does not support this my IP Phones will not be able to use the supplementary features such as Hol;d, Transfer, etc,etc.
I have escalated this to Cisc here in the UK for clarification. I am sure it will, but I just want clarification from Cisco or somebody that has used it in the same scenario.

"frags delayed" counter incrementing for Voice PVC

Hi,
We are using VoFR between two Cisco 2610 using FXO\FXS Cards. It is a point-point link with two PVCs, one for Voice and one for Data.
I have implemented Traffic-Shaping and FRF. However when i do a "show frame pvc " command, i can see "frags delayed"counter incrementing for the Voice PVC, indiciating delay in sending packets and thus compromising Voice Quality.
1. Is it normal to have this counter increasing ? What is the acceptable percentage i.e "frags delayed \ total frags" ?
2. Is there anything i can do ? Would PVC Priority Queuing help ?
I need to be sure if PVC Priority is the solution, as we would have to do a Flash Upgrade to install the new software with this feature.
++++++++++++++++++++++++++
show frame pvc 103
PVC Statistics for interface Serial0/0 (Frame Relay DTE)
DLCI = 103, DLCI USAGE = LOCAL, PVC STATUS = ACTIVE, INTERFACE = Serial0/0.3
input pkts 373951 output pkts 374604 in bytes 11542352
out bytes 12245392 dropped pkts 0 in FECN pkts 0
in BECN pkts 0 out FECN pkts 0 out BECN pkts 0
in DE pkts 0 out DE pkts 0
out bcast pkts 5474 out bcast bytes 1571038
pvc create time 11w3d, last time pvc status changed 04:06:31
Service type VoFR-cisco
Voice Queueing Stats: 0/100/0 (size/max/dropped)
Current fair queue configuration:
Discard Dynamic Reserved
threshold queue count queue count
64 16 2
Output queue size 0/max total 600/drops 0
configured voice bandwidth 30000, used voice bandwidth 0
fragment type VoFR-cisco fragment size 320
cir 32000 bc 320 be 0 limit 40 interval 10
mincir 32000 byte increment 40 BECN response no
frags 374604 bytes 12261814 frags delayed 6501 bytes delayed 1609296
shaping inactive
traffic shaping drops 0
+++++++++++++++++++++++++++++++++++

The following links explains the delay in voice traffic and gow to do traffic policing
VoIP over Frame Relay with QoS (Fragmentation, Traffic Shaping, LLQ / IP RTP Priority)
http://www.cisco.com/warp/public/788/voice-qos/voip-ov-fr-qos.html#15
Troubleshooting Output Drops with Priority Queueing
http://www.cisco.com/warp/public/105/priorityqueuedrops.html
Understanding Delay in Packet Voice Networks
http://www.cisco.com/warp/public/788/voip/delay-details.html
Voice QoS: ToS-CoS Mapping Via LLQ
http://www.cisco.com/warp/public/788/voice-qos/tos-cos.html
Frame Relay Traffic Shaping for VoIP and VoFR
http://www.cisco.com/warp/public/788/voip/fr_traffic.html

Darshan Prajapati -- About RTP support in J2ME

Hello friends I am developing the SIP client on J2ME. I want to route speech packets in real time in both client server direction. Also I want to continuously monitor the microphone to get the speech of the user and then simultaneously encode that speech and send it over wifi or internet through RTP packets.
So how can I do this? Please tell me api for getting continuous input from microphone(I do not want to first record and then send it.) and also making RTP packets to send it on internet.
Thanks in advance !
Darshan Prajapati

There is no RTP support in J2ME. Vikram Goyal has done some experiments using current mmapi but it does not work because of limitations of Manager and Player classes.
http://today.java.net/pub/a/today/2006/08/22/experiments-in-streaming-java-me.html?page=1

Help required in J2ME for developing software

Respected Friends,
I want to make an application in J2ME for Sony Ericson K700i mobile phone which is connected to the serial port of the PC,this application would be running inside the cellular mobile phone,when I will run this application all the files in the "File Manager" folder of mobile with .txt extensions would be copied to the PC to some paticular folder passing through the serial port to which the mobile is connected,I am facing some problems while devloping this application,my questions are as follows:
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2) Sony Ericson K700i does'nt implements JSR 75(PDA Optional Packages for the J2METM Platform,which is used to access PIM data and file systems) which means that we can not access the file system of this mobile so is it really possible to access the files stored in this mobile through my application?
3) Once the file is transferred to the serial port than will its contents of the file ramains in its original state or will be transferred to some kind of bianry data and we have to bring it in the original state.On the PC end will I need to write some other application which continiously monitores the serial port and will receive this file.
I would be waiting for your precious replies.
Thanking You,
Taqi Raza.

Define a plant/storage location for the delivery items which plant/SLoc combination is assigned to a warehouse (in data table T320 you can check plant/SLoc-WH assignments or SM30 V_T320.) In this case your delivery item should be relevant to WM-picking.
W/o warehouse management it is also possible to print picking list (you need a suitable output type for it for application "V2" (shipping)).
Edited by: Csaba Szommer on Jul 15, 2010 10:23 AM

AutoQoS for voice traffic settings?

Hi Everybody,
I have enabled auto qos on switch and following are information
Voice is the most important traffic in network, must ensure voice traffic goes first
SW# show mls qos map dscp-output-q
Dscp-outputq-threshold map:
d1 :d2 0 1 2 3 4 5 6 7 8 9
0 : 04-03 04-03 04-03 04-03 04-03 04-03 04-03 04-03 04-01 04-02
1 : 04-02 04-02 04-02 04-02 04-02 04-02 03-03 03-03 03-03 03-03
2 : 03-03 03-03 03-03 03-03 02-03 02-03 02-03 02-03 02-03 02-03
3 : 02-03 02-03 03-03 03-03 03-03 03-03 03-03 03-03 03-03 03-03
4 : 01-03 01-03 01-03 01-03 01-03 01-03 01-03 01-03 02-03 02-03
5 : 02-03 02-03 02-03 02-03 02-03 02-03 02-03 02-03 02-03 02-03
6 : 02-03 02-03 02-03 02-03
SW# show mls qos queue-set
Queueset: 1
Queue : 1 2 3 4
buffers : 10 10 26 54
threshold1: 138 138 36 20
threshold2: 138 138 77 50
reserved : 92 92 100 67
maximum : 138 400 318 400
For the
DSCP 46 : it's 01-03 (voice)
DSCP 0 : it's 04-03 (general traffic)
From my understanding
- 01-03 means queue 1 and threshold3. (by default threshold3 is 100 and hidden)
- queue-set 1 is enabled by default on all interface and hidden
According to the above information,
- Does the Auto Qos is design for voice goes first?
- Why the Q1 buffer and maximum are less then Q4? isn't suppose to set more buffer on Q1 for voice traffic? or I have to re-distribute the queue buffer and threshold, etc...
- or I just use priority-queue out, then those queue setting will be ignored?
Thanks in advance
Sam

udp ports 16384 to 32767 for rtp traffic
1720 tcp for control (h323 protocol)

IPhone 5s with hearing aids via Bluetooth. Need to use the phone for voice, what's the best distance that the mic will pick up your voice

I wear hearing aids that are made for Apple, the audio will go into the hearing aids via Bluetooth. I'll need to use my phone for voice only.. I drive for work and have a wok oh one too and both are synced to a bluetooh headset (LG). I'm trying to figure out how to talk on both phones ( use speakerphone on work phone?) without using a bluretooh headset. I don't know the range of the microphone on the IPHON, thinking about hanging it around my neck or put the phone into a stand/holder on the dash and use the speakerphone feature. Looking for ideas to make this simple

You can't do anything with that problem.
Take your iPhone to the closest Apple Store and they will give you new one for 200€/$ or if you bought your iPhone in less than 12 months you will get it for free.

Report for stock transfer from plant to plant

Hi,
Is there any standard report for stock transfer from plant to plant .
If not is there any enhancements to get the above report.
Regards,
Ravindranath.

Hi Ravi,
U can use the tcode MB51 and their u can give the input as 301 if it is 1 step in the mvt type or use 303 and 305 for 2 step transfer. and generate the report.
if u want the list of stock transport orders then use tcode ME2N and the input parameter as item category "U" and generate the report.
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Reward if useful
Shawn

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Attachments:
TCP_Serial_Host.vi ‏33 KB
TCP_Serial_RT.vi ‏41 KB

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Pending Quantity in PO for stock transfer

Dear All,
I want to know the pending order quantity for stock transfer orders (plant to plant),I have tried
ME80FN,ME2L Etc. but all report are showing the pending quantity only after MIGO done by
the receiving plant.
Our requirement is like this that one plant created one stock transfer order to one
plant and he want to know the actual pending order quantity as on date considering
by deducting stock in transit (5B5T) (i.e. PO quantity 100 pcs - quantity delivered 50 pcs then actual
pending quantity should be 50 pcs but in mentioned report it is showing 100 pcs pending).
Please note in our case in every PO more than 800 line items are there.
Can any one inform me about standard report or any table where net pending quantity are getting
stored.I have checked table EKBE,MARD,MARC.
Regards,
SP Shukla.

Hi,
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Pandari

Excise invoice for Stock Transfer from Mfg. Plant to excisable Depot

Hi Experts,
Which tcode (j1iin or j1is) should be used for creating excise invoice in mfg. plant for transfer the goods to exciseable depot and why.
And if we procure the material in mfg. plant and transferring the procured material to excisable depot then how can we map or assign incoming excise invoice to outgoing excise invoice so that same duty can be transferred.
Thanks in advance and points on satisfying answers.
Kuldeep Dahiya
Edited by: Kuldeep Dahiya on Jul 4, 2008 4:41 PM

Hi There
we will be using J1IIN for the transfer for the simple reason that you cannot move the goods from the manuefacturing plant without paying the excise duty.
regarding your second query I am unable to understand your exact requirement but as per my understanding you will do the good receipt and CENVAT will be credited and outgoing or stock transfer will happen as the regular process afterwards by debiting the excise duty from the CENVAT account
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Manu Seth

Proforma invoice not created for stock transfer order

Hi Experts
while creating a proforma invoice for stock transfer order between the plants we are not able to create proforma invoice , the process is replenishment delivery is created and Wm is also done and the billing with reference to delivery and the item category is relevant for billing with value J, and below is the message i get whilw creating invoice
     Technical data
         Tech. data details
          Client
          Group Number
          Sales Document Number
          Item Number of the SD Document         000010
          Schedule Line Number                   0001
          Counter in Control Tables              00
          Message Identification                 00
          System Message Number                  001
          Output Type                            E
          Message Variable 01
          Message Variable 02
          Message Variable 03
          Message Variable 04
          Group Type                             F
No biliing documents generated
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Thanks

This explanation is not in detail and hence I am not clear about your configuration settings.
First go to IMG Logistics-General => Tax on Goods Movements => India => Business Transactions => Outgoing Excise Invoices => Assign Billing Types to Delivery Types.
There ensure that you have assigned the billing type JEX (assuming that you are using the standard) to your delivery type NL.
Perhaps, you can also try with "D" for your item category in Billing Relevance field but I am not sure this will solve your issue. Try in development client.
thanks
G. Lakshmipathi

One E1 for voice and digital modem...

Hello All!!!
My question is about remote access and Ip-telephony ...
I use cisco3825 with PVDM2-12DM NM-1CE1T1-PRI cards installed.
I have only one E1 channel.
Can I install on router IOS with CCME support and use this E1 for voice too?
How I can detect what call should be forwarded to CME or Digital modem?
Thank you very much...

You can't do that with the hardware that you have now.
I recommend you get the following:
one VWIC or VWIC2, with two ports.
and
one HWIC-1CE1T1-PRI that supports the DMs,
and
one PVDM2-64. (you cannot use the -32 because you have two ports)
and
IOS 12.4(11)XW6
see:
http://cisco.com/en/US/prod/collateral/modules/ps2797/product_data_sheet0900aecd80710c88.html
and
http://cisco.com/en/US/docs/ios/12_4t/12_4t11/htpvdm2.html
and
http://cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00804794c6.shtml
One the VWIC one port you connect to telco. the other using with a crossed cable, you connect to the HWIC.
You route voice calls to the HWIC based on calling or called number. Automatic recognition of modem/voice calls is not supported.
Hope this helps, please rate post if it does!

RTP simulation on j2me for voice transfer

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