Is Voip over Umts disabled?

I have read in the Nokia support page that this phone support Voip. This is not the truth. My last phone was Nokia n95 8gb and whit this phone i can make a a voip call over wifi or over umts (simply configuring the built-in client, select the codec with voip settings and call). I have done the same with this 5630 but the voip built-in client doesn't work with umts. I can call over wifi but when i select a umts connection the message is always the same : sorry no connection available. This is false! The signal 3.5g says that the phone is connecting and then connected. I can surf with web browser (also opera mini) with my umts. Where is the problem? Probably Nokia have voip over umts disabled?
Another problem... when i make a videocall and i want use the first camera the software phone crash! During the call i select "use the other camera": the image remain 3 second suspended and then the phone restart...
What can i do?

I have solved my problem!!! :-) I noticed that nokia e52 is the same as mine!!!
There are a few steps to enable VOIP calling on the E52:
1. Download and install SIP VoIP 3.x Settingshttp://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin...
2. Create New SIP account in Ctrl Panel>Settings>Connection>SIP settings
3. Activate AWCDMA to allow VoIP calls over 3G network (not just WiFi)
a. Control Panels>Net Settings>Advanced VoIP settings>
b. select service you created in step 2 – eg pennytel
c. Scroll to "AWCDMA" and set it to "on"
4. Choose a contact from the normal Contacts menu, right scroll and choose Make Internet Call

Similar Messages

Solution: Gizmo VoIP over 3G/WCDMA with the Nokia ...

Copied from: http://www.elisanet.fi/~y639777/sipvoip
I have received some emails asking about how to get Nokia E-series working with Gizmo (SIP VoIP) over 3G/WCDMA.. So these are the steps that worked for me..
Steps:
1) first ensure that SIP VoIP is working over your wireless connection, ie following these instructions
2) disconnect from your Gizmo account if connected:
Nokia E51: Menu: Connectivity --> Internet tel. --> Options --> Disconnect from service
Nokia E71: Menu: Communications --> Internet tel. --> Options --> Disconnect from service
..then click "Back" and "Exit" until you get exit all the menus..
3) Find your default access point:
E51/E71: Menu: Tools --> Settings --> Connection --> SIP settings --> GIZMO VoIP (Not registered) --> Default access point
4) change you access point from your wireless network to 3G data connection, eg in my network the access point is called "internet".. then click "Back" and "Exit" until you get exit all the menus..
5) go back to the Gizmo account in Internet telephony:
Nokia E51: Menu: Connectivity --> Internet tel. --> Options --> Connect to service
Nokia E71: Menu: Communications --> Internet tel. --> Options --> Connect to service
6) Select your 3G data access point, it should be having a star and 3G label next to it
Notes:
- My wireless network is a simple configuration, basic ZyXEL ADSL box configured with NAT
- I know my operator doesn't block SIP VoIP traffic, additionally I checked this with a PC based SIP VoIP client
- My Nokia E-series device is not an operator variant. If you have an operator variant these features may be disabled in the firmware

Well I have found out that my 3G provider is not blocking ports. However, I have found that the issue only happens on that particular network, not others ..
I wonder if there are any setting I can change on my phone adapter that can help ..

No modem conection with voip over frame-relay

I'm trying to connect two modems through voip over frame-relay network using the modem passthrough command but that is no working. The voice calls is operating very well but the modems no connect between them. I'm using G-729 codec with ip header compression and data fragment in 80 bytes.

There are some know issues with cisco dsps and V.90+ modems, but we have been able to get most modems to work correctly accross our network. If you are using g729 for all other voice calls you will want to use the modem passthrough command to have both gw's turn to modem passthrough. I typically set up each dial peer with the specific commands, since I have found the global commands are not consistently applied. Here is a sample dial peer config....
dial-peer voice 3 voip
description Default voip peer
preference 1
incoming called-number .T
destination-pattern .T
progress_ind setup enable 3
progress_ind connect enable 8
modem passthrough nse codec g711ulaw
voice-class codec 1
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
fax rate disable
fax protocol pass-through g711ulaw
ip qos dscp cs3 signaling
no vad
Some of these commands are specific to the isdn signaling set up we have at this site. Also, we are using sip hence the sip session target. But the modem and fax passthrough commands are pretty much universal.
Also, some of those commands are not supported until 12.2.13. You might want to think about an ios upgrade.

VOIP over VPN need clarification

Hi,
Recently I have implemented Site-to-Site VPN between ASA and sonic wall firewall.
Problem: I can able to make call from ASA side(inside) Ip phone to sonic wall (inside) side Ip phone and vice versa and it’s ringing, But not able to hear voice. So I created VOIP over VPN configuration and applied appropriate service policy towards outside interface. But still I was not able to hear voice.
Tried below mentioned t’shot steps:
From ASA side we had two subnets (10.20.1.x/24 – Data and 10.20.2.x/24 – Voice ) and one subnet (192.168.x.x/24 ) from sonic wall side as interesting traffic ( lan to lan). When I configured site-to-site configuration on both ends my phase-1 and phase-2 came UP and can able to communicate between each other. (In interesting traffic I created two objects and bind those objects as one object-group for source i.e. ASA side lan subnet and one object for remote-Lan as destination)
My call manager is rest behind ASA and Ip phones needs to communicate from sonic wall side to inside ASA.
I can able to make call from ASA side(inside) Ip phone to sonic wall (inside) side Ip phone and vice versa and it’s ringing, But not able to hear voice. So I created VOIP over VPN configuration and applied appropriate service policy towards outside interface. But still I was not able to hear voice.
So, I done supernetting the data subnet and voice subnet into single network i.e. 10.20.x.x/16 at ASA side and applied the configuration changes (changed ACL, nonat rule, Voice QOS ACL accordingly), and I’m able to hear voice both end and I can communicate properly from ASA inside Ip phone to Sonic wall inside Ip phone and vice versa.
My question: I’m not understanding the logic how this supernetting resolved dead voice issue.
Pls clarify my question I’m bit confused on this.

It's not recommended. Although VPNs guarantee a secure pipe end-to-end, they don't guarantee latency and variations in latency (Jitter).

How to implement VoIP over SIP in J2ME

Hi All,
This is kapil from India and i want to implement VoIP over SIP in j2me.Any help will be appreciated.
I know about SIP API in J2ME. I want to use VoIP over it.Please help.I will be grateful to you.
regards
kapil

Why don't you give us the money your Boss pays you.. then we'll see what can be done ;)
Seriously, it might be smart to start your own research first. Check out what VoiP and SIP are all about, and the check how it fits into J2me.

VOIP over broadband - with QoS?

At the moment we use Cisco 2600 routers with 256k BT kilostream connections to our stations, to provide IT and Cisco VOIP. The kilostream connections cost about ?5k/station/year.
We want to stream video for the IT, but the kilostream isn't up to it. We also have BT broadband (20:1 contention), to the station, generally between 2-8M, which costs about ?500/station/year. We can do video over that fine.
My questions are:-
1) I'd like to get rid of the expensive kilostream and just use broadband. However, we have about 10-25 VOIP phones on each station, and we keep being told that broadband could not support that, the latency is too high, the reliability too low, that it needs QoS and you can't get that with this infrastructure etc etc. Can I use Cisco VOIP over broadband to 2600 routers, would QoS work and what about the quality/reliability? If it is possible, any hints as to how?
2) As an alternative, I was wondering if I could have both kilostream and broadband connected to the back of my 2600 and have different services going over the two connections (eg voice over kilostream and IT over broadband)? And if so, maybe I could have failover from one to another? Can I do this, and if so, any ideas how?
I don't know if it helps, but we normally terminate our broadband IPSEC VPNs on our WatchGuard firewall, although we do also have a Microsoft ISA firewall that we could use instead.
Any help you are able to offer would be very gratefully received.
Regards
Eric

At the moment we use Cisco 2600 routers with 256k BT kilostream connections to our stations, to provide IT and Cisco VOIP. The kilostream connections cost about ?5k/station/year.
We want to stream video for the IT, but the kilostream isn't up to it. We also have BT broadband (20:1 contention), to the station, generally between 2-8M, which costs about ?500/station/year. We can do video over that fine.
My questions are:-
1) I'd like to get rid of the expensive kilostream and just use broadband. However, we have about 10-25 VOIP phones on each station, and we keep being told that broadband could not support that, the latency is too high, the reliability too low, that it needs QoS and you can't get that with this infrastructure etc etc. Can I use Cisco VOIP over broadband to 2600 routers, would QoS work and what about the quality/reliability? If it is possible, any hints as to how?
2) As an alternative, I was wondering if I could have both kilostream and broadband connected to the back of my 2600 and have different services going over the two connections (eg voice over kilostream and IT over broadband)? And if so, maybe I could have failover from one to another? Can I do this, and if so, any ideas how?
I don't know if it helps, but we normally terminate our broadband IPSEC VPNs on our WatchGuard firewall, although we do also have a Microsoft ISA firewall that we could use instead.
Any help you are able to offer would be very gratefully received.
Regards
Eric

DiffServ & TE paths are required to deploy commercial VoIP over a MPLS net.

Dear members.
There's something I'd like to know from those that truly and successfully
implement commercial VoIP over a MPLS infrastructure.
Supporting IP QoS (DiffServ) across the core backbone and/or MPLS traffic-
engineered paths is really required ?
Consider a whole backbone built with high speed throughput connections, plenty of bandwidth availability and no periods of congestion!
I do believe TE tunnels (paths) are necessary for quick recovery when a node
or link fails, but rather them assuming something I haven't experienced yet
I'd like to listen from those who really have already successfully deployed
commercial VoIP.
Best regards.
Murilo Pugliese.

With increasing adoption of voice over IP (VoIP), the landscape for deployment is rapidly changing. Service providers are often driven by the need to provide customers a high grade of service to carry voice traffic across a network. However, today's multiservice packet networks rely on IP-based packet switching. In addition, IP by itself is simply best-effort service that is not sufficient to provide the strict delay, jitter, and bandwidth guarantees required for VoIP and other real-time traffic. Cisco IOS QoS features are ideal for this situation. Using the IETF differentiated services (DiffServ) model for QoS, VoIP traffic can be treated appropriately.
http://www.cisco.com/en/US/netsol/ns341/ns396/ns172/ns155/networking_solutions_white_paper09186a00800a8441.shtml

Adding a picture over a disabled JButton

Hello,
I am trying to add an image over a disabled JButton.
The code I have so far is:
buttons[i][j].setEnabled(false);
ImageIcon mine = new ImageIcon("images/mine.gif");
buttons[i][j].setIcon(mine);This correctly adds the mine over the button, but the color disappear and the image is grayish. I am trying to keep the initial colors for the image (that appears on the JButton).
Anyone has any idea please?
Thank you very much.

Let me guess: you're creating a Minesweeper app. If so let me tell you what I did: Each "cell" held a JPanel that used the CardLayout manager. Initially the JPanel showed a JButton. Right clicking the button could change its icon (where you "flag" a button that you think holds a mine). Clicking a button will cause the CardLayout to swap the button for a JLabel that either showed a number, or was blank, or showed an image of a blown-up mine.
The other thing I did was to try to separate the logic of the Cell from the GUI portion -- to use an MVC pattern. It might have been overkill for such a simple application, but it taught me a lot about MVC and was a lot of fun to create. Best of luck and HTH.

N95 - VOIP / Internet Telephony disabled - Network...

I have a Vodaphone n95 where the Internet Telephony feature has been disabled which is a real bummer as i cant use gizmo at all.
I called Voda and they say, its a company decision to remove it as VOIP is not in use on Vodaphone.
Wasnt really happy with that and spent all day yesterday looking for a way to upgrade my firmware and found a site on the net where i found out some interesting info to do with VOIP, specifically this post -
http://www.mobilevoipforum.org/showthread.php?t=515
Its a bit shabby that Vodaphone have done this to my phone, cause I used my gizmo account with my e61 with no problems over WIFI.
bixby
e61 and loving it !

!!!!!!!!!! PEOPLE IF YOUR ANNOYED, USE YOUR BRAIN AND RETURN IT BACK TO THE NETWORK USING THE BELOW EMAIL AS YOUR BASIS !!!!!!!!!
Dear Mr. XXXXXXX,
Thank you for your enquiry to Consumer Direct date 23/04/07
Your reference number for this case is XXXXXXXXX and should be quoted in all further correspondence regarding this case.
When you entered into contract with the trader for this phone, it should clearly correspond to it’s description. Therefore, if there is a feature which has been removed, you could argue that phone is no longer as described. This may be considered to be a breach of contract under the Sale of Goods Act 1979 (as amended).
We would therefore recommend that you return to the company who sold you the phone and request that this feature is provided according to it’s description. Alternatively, you could negotiate an alternative replacement phone which suits you or request compensation as a result of this breach.
Your contract with the service provider for the performance of the contract may be treated as a separate contract. Therefore, this should not be affected as it would appear that the problem is only with the handset in this instance.
We will also pass the details of this enquiry to Trading Standards.
If you require any further advice or information about this case, please do not hesitate in contacting Consumer Direct on 08454 04 05 06 quoting your reference number
Hope this helpsMessage Edited by bixby on 30-Apr-200708:31 AM
e61 and loving it !

VOIP over VPN dropp RTP protocol

We are installing a new 2911 ISR in our office and connecting with a Linksys (CISCO) RV016 VPN router. These are two small doctors offices that need to have computer, and Voip traffic over a VPN.
Currently we connect an older RV082 and the RV016 together and have NO issues with VOIP traffic. If we establish a connection with the 2911 router then we are having an issue with no voice or RTP traffic coming through. Phones will connect, and dial out, but no voice can be heard.
The First office is on a Verizon Fios Network with a MTU of 1492. The Network and servers are as follows:
Remote Office                                                                            Main Office
Linksys Spa 942 phones
               |
Netgear 10/100 POE Switch
               |                                                                        =================
Linksys (cisco) RV016 VPN                                            |          2911                 |
               |                                                                       |    POE Module Sw     |
Comcast Cable Modem                                                  -------------------------------
               |                                                                                       |          |
             VPN                                                                                VPN       |
               +=======================================+     Asterisk
                                                                                                              (Call Man)
Basically we have the Internet coming in from Gig0/0 and routing traffic to multiple outside IP addresses so we are using 3 subs in our configuration.
192.168.1.X          192.168.2.X          192.168.3.X               192.168.0.X (Remote Group)
When we connect the old routers (RV016 and RV082) VPN VOIP and Data traffic go fine.   We are using a Term Server on one end, Web Server, and the Asterisk PBX for our VOIP Call Manager.
So far we connect up the 2911 and the RV016 and have no issues with data traffic. But the VOIP is dead on the remote end. No sound. We did a Wireshark on traffic, and we are getting some 407 errors from the Astersick Host, and a unknown RTP version 1 error message. THe only thing that we had to do on the RV082 router was port forward UDP 506 and 10001 - 20000 for the traffic, and setup a access rule, but nothing else.
We are getting traffic on the 2911, but nothing else. We have excluded the 192.158.0.X traffic from the NAT so not to get into that issue, and have even tried forwarding ports but nothing seems to help. Is there a good way to route this traffic? Our bandwidth is pretty fast so I am not sure if QoS is needed, but if so it is not one of my strong areas. What is the best way to route this traffic through the VPN without loosing the RTP part of the call.

I put this line in and still not getting audio on the other end. I will be doing captures tonight from working and non working phones. I need to get this resolved. I have spent 3 weeks on this issue and I have run out of time. Should I use the DEBUG VOIP SIP command for the capture on the router? I believe this would be the best resolution to the service to see what is going on. The phones work with a RV016 and RV082 router in place. All data traffic works fine in sending and recieving calls.
I have read about all of the articles on Cisco and voip traffic. We are going to be shutting off the natting on the router to see if I can just get the voip traffic to flow. Once we get it flowing then I can work on building up the house on a stable foundation.
At this time, we are routing multiple IP addresses throught the 2911 and have IP NAT OUTSIDE on the G0/0 port and IP NAT INSIDE on the G1/0 Interface, which is a POE Switch Module in the 2911.
I know that the cisco router wants to act as a call manager, or terminate the SIP traffic on the 2911, but we have a working Asterisk box that handles all SIP traffic. If there is a way to just forward the traffic there properly, without the 2911 trying to intercept the traffic, that would be wonderful. I am looking at the possibility of creating dial-peer groups for all of the phones, but really is this needed? What is so frustrating about the whole situation is that I put in a 5 year old sub $200 router and everything works.
Dale

VoIP over VPN won't work

Hi,
I have a Cisco ASA 5540 running 8.2(5). When I dial a phone on the other of the the VPN the first time I get a blank after it rings(i.e when the voice mail get activated ot if someone picks the phone up), however works the second and consequent times i dial.
A little background.
Two sites A and B connected via IPsec Tunnel. No problems in communication except for the VoIP issue.
A Phone in on site A(172.17.168.x) and other on site B(192.168.103.x)
SiteA and SiteB is connected via an IPsec tunnel on the Cisco ASA.
First call fails.
Second call works.
Result of a packet trace is also the same. The UDP packet get drops when tried for the first time but subsequent ones pass.
Any ideas on how to fix this?
First time
ASA5520# packet-tracer input inside udp 172.17.168.95 10000 192.168.3.103 10000
Phase: 1
Type: ROUTE-LOOKUP
Subtype: input
Result: ALLOW
Config:
Additional Information:
in   0.0.0.0         0.0.0.0         outside
Phase: 2
Type: IP-OPTIONS
Subtype:
Result: ALLOW
Config:
Additional Information:
Forward Flow based lookup yields rule:
in id=0xb13db840, priority=0, domain=inspect-ip-options, deny=true
hits=2603604, user_data=0x0, cs_id=0x0, reverse, flags=0x0, protocol=0
src ip=0.0.0.0, mask=0.0.0.0, port=0
dst ip=0.0.0.0, mask=0.0.0.0, port=0, dscp=0x0
Phase: 3
Type: NAT-EXEMPT
Subtype:
Result: ALLOW
Config:
match ip inside 172.17.168.0 255.255.255.0 outside 192.168.0.0 255.255.0.0
    NAT exempt
    translate_hits = 141, untranslate_hits = 6
Additional Information:
Forward Flow based lookup yields rule:
in id=0xb13eccf0, priority=6, domain=nat-exempt, deny=false
hits=323, user_data=0xb13ecc30, cs_id=0x0, use_real_addr, flags=0x0, protocol=0
src ip=172.17.168.0, mask=255.255.255.0, port=0
dst ip=192.168.0.0, mask=255.255.0.0, port=0, dscp=0x0
Phase: 4
Type: NAT
Subtype:
Result: ALLOW
Config:
nat (inside) 1 0.0.0.0 0.0.0.0
match ip inside any outside any
    dynamic translation to pool 1 (Public IP)
    translate_hits = 1005381, untranslate_hits = 37221
Additional Information:
Forward Flow based lookup yields rule:
in id=0xb1490078, priority=1, domain=nat, deny=false
hits=1805592, user_data=0xb148ffb8, cs_id=0x0, flags=0x0, protocol=0
src ip=0.0.0.0, mask=0.0.0.0, port=0
dst ip=0.0.0.0, mask=0.0.0.0, port=0, dscp=0x0
Phase: 5
Type: NAT
Subtype: host-limits
Result: ALLOW
Config:
nat (inside) 1 0.0.0.0 0.0.0.0
match ip inside any outside any
    dynamic translation to pool 1 (Public IP)
    translate_hits = 1005392, untranslate_hits = 37221
Additional Information:
Forward Flow based lookup yields rule:
in id=0xb14903b0, priority=1, domain=host, deny=false
hits=2728036, user_data=0xb148ffb8, cs_id=0x0, reverse, flags=0x0, protocol=0
src ip=0.0.0.0, mask=0.0.0.0, port=0
dst ip=0.0.0.0, mask=0.0.0.0, port=0, dscp=0x0
Phase: 6
Type: VPN
Subtype: encrypt
Result: DROP
Config:
Additional Information:
Forward Flow based lookup yields rule:
out id=0xb1fbe6a0, priority=70, domain=encrypt, deny=false
hits=106, user_data=0x0, cs_id=0xb1fa5bc8, reverse, flags=0x0, protocol=0
src ip=172.17.168.0, mask=255.255.255.0, port=0
dst ip=192.168.0.0, mask=255.255.0.0, port=0, dscp=0x0
Result:
input-interface: inside
input-status: up
input-line-status: up
output-interface: outside
output-status: up
output-line-status: up
Action: drop
Drop-reason: (acl-drop) Flow is denied by configured rule
Second time
ASA5520# packet-tracer input inside udp 172.17.168.95 10000 192.168.3.103 10000
Phase: 1
Type: ROUTE-LOOKUP
Subtype: input
Result: ALLOW
Config:
Additional Information:
in   0.0.0.0         0.0.0.0         outside
Phase: 2
Type: IP-OPTIONS
Subtype:
Result: ALLOW
Config:
Additional Information:
Forward Flow based lookup yields rule:
in id=0xb13db840, priority=0, domain=inspect-ip-options, deny=true
hits=2603850, user_data=0x0, cs_id=0x0, reverse, flags=0x0, protocol=0
src ip=0.0.0.0, mask=0.0.0.0, port=0
dst ip=0.0.0.0, mask=0.0.0.0, port=0, dscp=0x0
Phase: 3
Type: NAT-EXEMPT
Subtype:
Result: ALLOW
Config:
match ip inside 172.17.168.0 255.255.255.0 outside 192.168.0.0 255.255.0.0
    NAT exempt
    translate_hits = 142, untranslate_hits = 6
Additional Information:
Forward Flow based lookup yields rule:
in id=0xb13eccf0, priority=6, domain=nat-exempt, deny=false
hits=324, user_data=0xb13ecc30, cs_id=0x0, use_real_addr, flags=0x0, protocol=0
src ip=172.17.168.0, mask=255.255.255.0, port=0
dst ip=192.168.0.0, mask=255.255.0.0, port=0, dscp=0x0
Phase: 4
Type: NAT
Subtype:
Result: ALLOW
Config:
nat (inside) 1 0.0.0.0 0.0.0.0
match ip inside any outside any
    dynamic translation to pool 1 (Public IP)
    translate_hits = 1005470, untranslate_hits = 37222
Additional Information:
Forward Flow based lookup yields rule:
in id=0xb1490078, priority=1, domain=nat, deny=false
hits=1805881, user_data=0xb148ffb8, cs_id=0x0, flags=0x0, protocol=0
src ip=0.0.0.0, mask=0.0.0.0, port=0
dst ip=0.0.0.0, mask=0.0.0.0, port=0, dscp=0x0
Phase: 5
Type: NAT
Subtype: host-limits
Result: ALLOW
Config:
nat (inside) 1 0.0.0.0 0.0.0.0
match ip inside any outside any
    dynamic translation to pool 1 (Public IP)
    translate_hits = 1005479, untranslate_hits = 37223
Additional Information:
Forward Flow based lookup yields rule:
in id=0xb14903b0, priority=1, domain=host, deny=false
hits=2728365, user_data=0xb148ffb8, cs_id=0x0, reverse, flags=0x0, protocol=0
src ip=0.0.0.0, mask=0.0.0.0, port=0
dst ip=0.0.0.0, mask=0.0.0.0, port=0, dscp=0x0
Phase: 6
Type: VPN
Subtype: encrypt
Result: ALLOW
Config:
Additional Information:
Forward Flow based lookup yields rule:
out id=0xb23a5060, priority=70, domain=encrypt, deny=false
hits=1, user_data=0x202004, cs_id=0xb1fa5bc8, reverse, flags=0x0, protocol=0
src ip=172.17.168.0, mask=255.255.255.0, port=0
dst ip=192.168.0.0, mask=255.255.0.0, port=0, dscp=0x0
Phase: 7
Type: VPN
Subtype: ipsec-tunnel-flow
Result: ALLOW
Config:
Additional Information:
Reverse Flow based lookup yields rule:
in id=0xb237e278, priority=69, domain=ipsec-tunnel-flow, deny=false
hits=1, user_data=0x2040e4, cs_id=0x0, reverse, flags=0x0, protocol=0
src ip=192.168.0.0, mask=255.255.0.0, port=0
dst ip=172.17.168.0, mask=255.255.255.0, port=0, dscp=0x0
Phase: 8
Type: IP-OPTIONS
Subtype:
Result: ALLOW
Config:
Additional Information:
Reverse Flow based lookup yields rule:
in id=0xb138d7e0, priority=0, domain=inspect-ip-options, deny=true
hits=2440659, user_data=0x0, cs_id=0x0, reverse, flags=0x0, protocol=0
src ip=0.0.0.0, mask=0.0.0.0, port=0
dst ip=0.0.0.0, mask=0.0.0.0, port=0, dscp=0x0
Phase: 9
Type: FLOW-CREATION
Subtype:
Result: ALLOW
Config:
Additional Information:
New flow created with id 2676819, packet dispatched to next module
Module information for forward flow ...
snp_fp_tracer_drop
snp_fp_inspect_ip_options
snp_fp_adjacency
snp_fp_encrypt
snp_fp_fragment
snp_ifc_stat
Module information for reverse flow ...
snp_fp_tracer_drop
snp_fp_inspect_ip_options
snp_fp_ipsec_tunnel_flow
snp_fp_adjacency
snp_fp_fragment
snp_ifc_stat
Result:
input-interface: inside
input-status: up
input-line-status: up
output-interface: outside
output-status: up
output-line-status: up
Action: allow

No I haven't, but I'll call them today. That doesnt explain the VPN stopped working though, and at this point I'd rather have L2TP/IPSec working to my box at home, rather than an unencrypted session running over port 445. Appreciate the suggestion though...

Is Low MTU (ie 320) for VoIP over Frame-relay can be used to avoid serialization delay for large data packets?

In order to provide voice quality for VoIP, is FR fragmentation equivalent to lowering the MTU size for the serial subinterface of a Frame relay subinterface?
Is there any isues like stop communications, using Low MTU ?

If the router belongs to any of the
platform listed below, then use
FRF12 for you fragmentation. The MTU size
lowering also works but this can cause a high over
head as it can't be specified on a per dlci level.
With multiple dlci i.e. subinterfaces use per dlci fragmentation.. this helps reduce the overhead of changing the MTU size of the physical interface.
Snip config example.( __Must be configured on both side of the termination___)
PHONE 3333312---ROUTERA ----DLCI 100----ROUTERB ---PHONE 2111123
ROUTER A
dial-peer voice 1 voip
destination-pettern 2T
session target ipv4:10.10.10.2
int ser 0/0
encap frame-relay
frame-relay traffic-shaping
no ip address
interface serial0/0.1 point-to-point
ip add 10.10.10.1 255.255.255.252
frame-relay interface-dlci 100
class voice
map-class frame-relay voice
frame cir 64000
frame bc 640
frame mincir 64000
frame-relay ip rtp priority 16384 16383 48
frame fragment 80
frame fair-queue 64 256 0
ROUTER B
dial-peer voice 3 voip
destination-pettern 3T
session target ipv4:10.10.10.1
int ser 0/0
encap frame-relay
frame-relay traffic-shaping
no ip address
interface serial0/0.1 point-to-point
ip add 10.10.10.2 255.255.255.252
frame-relay interface-dlci 100
class voice
map-class frame-relay voice
frame cir 64000
frame bc 640
frame mincir 64000
frame-relay ip rtp priority 16384 16383 48
frame fragment 80
frame fair-queue 64 256 0
This should help if your router is
c2600, c3600, mc3810, c7200, c1750
all running the right level of IOS.
12.1(5)T and above should work well.

VoIP over satellite link

I configured VoIP on a MPPP with LLQ as a QoS policy in a lab environment, it works fine.
Now I`d like to configure it over a satellite link with 128 Kbps of bandwidth and four channel.
What considerations should I to do in my configurations for it?
Are there some command configs for tunning the QoS in this environment?
I know that the delay between two points through satellite link is 500ms.
Someone has some experience with this case?. I`ll appreciate your suggestions.
Thank you.

hi everybody,
i have an internet connection via sat with the following config:
1. 2Mbps down and 1Mbps up.
2. ps4500 sat modem.
3. 2600XM router.
4. AS5400 gateway.
5. its connected to PSTN.
we have about 100 users that use it for calling through PSTN.
Now we like to conffigure QoS/Diffserv for Voice connection and connect about 200 users for normal internet connection.
can anybody tell me what the consumming B/W for 100 calls and what is the best configuration for this senario.
best redards
eng thaar
oil ministry

HTTP Post of big files over UMTS stops

Hi there I modified the code from [this thread|http://forum.java.sun.com/thread.jspa?forumID=256&threadID=451245] to suit the needs of J2ME:
package some.package.util.http;
* Original source code: http://forum.java.sun.com/thread.jspa?forumID=256&threadID=451245
* This source was adapted to meet the requirements of J2ME
* @author tb
import java.io.*;
import java.util.Enumeration;
import java.util.Vector;
import javax.microedition.io.Connector;
import javax.microedition.io.HttpConnection;
import javax.microedition.io.file.FileConnection;
* <code>MultiPartFormOutputStream</code> is used to write
* "multipart/form-data" to a <code>javax.microedition.io.HttpConnection</code> for
* POSTing. This is primarily for file uploading to HTTP servers.
public class MultiPartFormOutputStream {
* The line end characters.
private static final String NEWLINE = "\r\n";
* The boundary prefix.
private static final String PREFIX = "--";
* The output stream to write to.
private DataOutputStream out = null;
* The multipart boundary string.
private String boundary = null;
private Vector progressListeners;
* Creates a new <code>MultiPartFormOutputStream</code> object using
* the specified output stream and boundary. The boundary is required
* to be created before using this method, as described in the
* description for the <code>getContentType(String)</code> method.
* The boundary is only checked for <code>null</code> or empty string,
* but it is recommended to be at least 6 characters. (Or use the
* static createBoundary() method to create one.)
* @param os the output stream
* @param boundary the boundary
* @see #createBoundary()
* @see #getContentType(String)
public MultiPartFormOutputStream(OutputStream os, String boundary) {
if(os == null) {
throw new IllegalArgumentException("Output stream is required.");
if(boundary == null || boundary.length() == 0) {
throw new IllegalArgumentException("Boundary stream is required.");
this.out = new DataOutputStream(os);
this.boundary = boundary;
this.progressListeners=new Vector();
* Writes an boolean field value.
* @param name the field name (required)
* @param value the field value
* @throws java.io.IOException on input/output errors
public void writeField(String name, boolean value)
throws java.io.IOException {
writeField(name, new Boolean(value).toString());
* Writes an double field value.
* @param name the field name (required)
* @param value the field value
* @throws java.io.IOException on input/output errors
public void writeField(String name, double value)
throws java.io.IOException {
writeField(name, Double.toString(value));
* Writes an float field value.
* @param name the field name (required)
* @param value the field value
* @throws java.io.IOException on input/output errors
public void writeField(String name, float value)
throws java.io.IOException {
writeField(name, Float.toString(value));
* Writes an long field value.
* @param name the field name (required)
* @param value the field value
* @throws java.io.IOException on input/output errors
public void writeField(String name, long value)
throws java.io.IOException {
writeField(name, Long.toString(value));
* Writes an int field value.
* @param name the field name (required)
* @param value the field value
* @throws java.io.IOException on input/output errors
public void writeField(String name, int value)
throws java.io.IOException {
writeField(name, Integer.toString(value));
* Writes an short field value.
* @param name the field name (required)
* @param value the field value
* @throws java.io.IOException on input/output errors
public void writeField(String name, short value)
throws java.io.IOException {
Short val = new Short(value);
writeField(name, val.toString());
* Writes an char field value.
* @param name the field name (required)
* @param value the field value
* @throws java.io.IOException on input/output errors
public void writeField(String name, char value)
throws java.io.IOException {
writeField(name, new Character(value).toString());
* Writes an string field value. If the value is null, an empty string
* is sent ("").
* @param name the field name (required)
* @param value the field value
* @throws java.io.IOException on input/output errors
public void writeField(String fieldName, String value)
throws java.io.IOException {
if(fieldName == null) {
throw new IllegalArgumentException("name cannot be null or empty.");
if(value == null) {
value = "";
String contentDisposition = "Content-Disposition: form-data; name=\"" + fieldName + "\"";
--boundary\r\n
Content-Disposition: form-data; name="<fieldName>"\r\n
\r\n
<value>\r\n
// write boundary
out.write(PREFIX.getBytes());
out.write(boundary.getBytes());
out.write(NEWLINE.getBytes());
// write content header
out.write(contentDisposition.getBytes());
out.write(NEWLINE.getBytes());
out.write(NEWLINE.getBytes());
// write content
out.write(value.getBytes());
out.write(NEWLINE.getBytes());
//out.flush();
* Writes a file's contents. If the file is null, does not exists, or
* is a directory, a <code>java.lang.IllegalArgumentException</code>
* will be thrown.
* @param name the field name
* @param mimeType the file content type (optional, recommended)
* @param file the file (the file must exist)
* @throws java.io.IOException on input/output errors
public void writeFile(String fieldName, String mimeType, FileConnection fileCon)
throws java.io.IOException {
if(fileCon == null) {
throw new IllegalArgumentException("File cannot be null.");
if(!fileCon.exists()) {
throw new IllegalArgumentException("File does not exist.");
if(fileCon.isDirectory()) {
throw new IllegalArgumentException("File cannot be a directory.");
writeFile(fieldName, mimeType, fileCon.getPath(), fileCon.openDataInputStream());
* Writes a input stream's contents. If the input stream is null, a
* <code>java.lang.IllegalArgumentException</code> will be thrown.
* @param name the field name
* @param mimeType the file content type (optional, recommended)
* @param fileName the file name (required)
* @param is the input stream
* @throws java.io.IOException on input/output errors
public void writeFile(String fieldName, String mimeType,
String fileName, InputStream is)
throws java.io.IOException {
if(is == null) {
throw new IllegalArgumentException("Input stream cannot be null.");
if(fileName == null || fileName.length() == 0) {
throw new IllegalArgumentException("File name cannot be null or empty.");
String contentDisposition = "Content-Disposition: form-data; name=\"" + fieldName +
"\"; filename=\"" + fileName + "\"";
String contentType = "Content-Type: " + mimeType;
--boundary\r\n
Content-Disposition: form-data; name="<fieldName>"; filename="<filename>"\r\n
Content-Type: <mime-type>\r\n
\r\n
<file-data>\r\n
// write boundary
out.write(PREFIX.getBytes());
out.write(boundary.getBytes());
out.write(NEWLINE.getBytes());
// write content header
out.write(contentDisposition.getBytes());
out.write(NEWLINE.getBytes());
if(mimeType != null) {
out.write(contentType.getBytes());
out.write(NEWLINE.getBytes());
out.write(NEWLINE.getBytes());
// write content
byte[] data = new byte[1024];
int r = 0;
while((r = is.read(data, 0, data.length)) != -1) {
out.write(data, 0, r);
this.updateProgressListeners(data.length);
// close input stream, but ignore any possible exception for it
try {
is.close();
} catch(Exception e) { e.printStackTrace();}
out.write(NEWLINE.getBytes());
//out.flush();
* Writes the given bytes. The bytes are assumed to be the contents
* of a file, and will be sent as such. If the data is null, a
* <code>java.lang.IllegalArgumentException</code> will be thrown.
* @param name the field name
* @param mimeType the file content type (optional, recommended)
* @param fileName the file name (required)
* @param data the file data
* @throws java.io.IOException on input/output errors
public void writeFile(String fieldName, String mimeType, String fileName, byte[] data)
throws java.io.IOException {
if(data == null) {
throw new IllegalArgumentException("Data cannot be null.");
if(fileName == null || fileName.length() == 0) {
throw new IllegalArgumentException("File name cannot be null or empty.");
String contentDisposition = "Content-Disposition: form-data; name=\"" + fieldName +
"\"; filename=\"" + fileName + "\"";
String contentType = "Content-Type: " + mimeType;
--boundary\r\n
Content-Disposition: form-data; name="<fieldName>"; filename="<filename>"\r\n
Content-Type: <mime-type>\r\n
\r\n
<file-data>\r\n
// write boundary
out.write(PREFIX.getBytes());
out.write(boundary.getBytes());
out.write(NEWLINE.getBytes());
// write content header
out.write(contentDisposition.getBytes());
out.write(NEWLINE.getBytes());
if(mimeType != null) {
out.write(contentType.getBytes());
out.write(NEWLINE.getBytes());
out.write(NEWLINE.getBytes());
// write content
out.write(data, 0, data.length);
out.write(NEWLINE.getBytes());
//out.flush();
* Flushes the stream. Actually, this method does nothing, as the only
* write methods are highly specialized and automatically flush.
* @throws java.io.IOException on input/output errors
public void flush() throws java.io.IOException {
out.flush();
* Closes the stream. 
* 
* NOTE: This method MUST be called to finalize the
* multipart stream.
* @throws java.io.IOException on input/output errors
public void close() throws java.io.IOException {
// write final boundary
out.write(PREFIX.getBytes());
out.write(boundary.getBytes());
out.write(PREFIX.getBytes());
out.write(NEWLINE.getBytes());
out.flush();
out.close();
* Gets the multipart boundary string being used by this stream.
* @return the boundary
public String getBoundary() {
return this.boundary;
* Creates a new <code>javax.microedition.io.HttpConnection</code> object from the
* specified <code>String</code>.
* @return a <code>javax.microedition.io.HttpConnection</code> object for the URL
* @throws java.io.IOException on input/output errors
public static HttpConnection createConnection(String url)
throws java.io.IOException {
HttpConnection conn = (HttpConnection) Connector.open(url);
conn.setRequestMethod(HttpConnection.POST);
return conn;
* Creates a multipart boundary string by concatenating 20 hyphens (-)
* and the hexadecimal (base-16) representation of the current time in
* milliseconds.
* @return a multipart boundary string
* @see #getContentType(String)
public static String createBoundary() {
return "--------------------" +
Long.toString(System.currentTimeMillis(), 16);
* Gets the content type string suitable for the
* <code>javax.microedition.io.HttpConnection</code> which includes the multipart
* boundary string. 
* 
* This method is static because, due to the nature of the
* <code>javax.microedition.io.HttpConnection</code> class, once the output stream
* for the connection is acquired, it's too late to set the content
* type (or any other request parameter). So one has to create a
* multipart boundary string first before using this class, such as
* with the <code>createBoundary()</code> method.
* @param boundary the boundary string
* @return the content type string
* @see #createBoundary()
public static String getContentType(String boundary) {
return "multipart/form-data; boundary=" + boundary;
public Enumeration getProgressListeners() {
return progressListeners.elements();
public void addProgressListener(ProgressListener pl) {
this.progressListeners.addElement(pl);
private void updateProgressListeners(int length) {
for(int i = 0;i<this.progressListeners.size();i++)
ProgressListener p = (ProgressListener) this.progressListeners.elementAt(i);
try {
p.updateProgress(length);
}catch(IllegalArgumentException ex)
ex.printStackTrace();
private void initProgressListeners(int maxValue)
for(int i = 0;i<this.progressListeners.size();i++)
ProgressListener p = (ProgressListener) this.progressListeners.elementAt(i);
p.setMaximum(maxValue);
{code}
Modifications I made:
* Use javax.microedition.io.HttpConnection instead of URLConnection
* Use String.getBytes() as DataOutputStream.writeBytes(String s) is not available in MIDP
* Use FileConnection from JSR-75 FileConnection API
I use it like this:
{code:java}
String boundary = MultiPartFormOutputStream.createBoundary();
// this.httpCon.setRequestProperty("Accept", "*/*");
this.httpCon.setRequestMethod(HttpConnection.POST);
this.httpCon.setRequestProperty("Content-Type", MultiPartFormOutputStream.getContentType(boundary));
this.httpCon.setRequestProperty("Connection", "Close");
// this.httpCon.setRequestProperty("Cache-Control", "no-cache");
this.multipartOutput = new MultiPartFormOutputStream(this.httpCon.openDataOutputStream(), boundary);
if(this.progressListener!=null)
this.multipartOutput.addProgressListener(progressListener);
this.multipartOutput.writeField("hwid", "");
this.multipartOutput.writeField("lang", "de-de");
this.multipartOutput.writeField("guilang","en-gb");
this.multipartOutput.writeField("hwid", "0000000DECAF");
this.multipartOutput.writeField("actcode", "masterKey");
this.multipartOutput.writeField("x1", "-1");
this.multipartOutput.writeField("y1", "-1");
this.multipartOutput.writeField("x2", "-1");
this.multipartOutput.writeField("y2", "-1");
this.multipartOutput.writeFile("img", "image/jpeg", "testbild.jpg", dataInputStream);
this.multipartOutput.close();
this.httpCon.close();
{code}
While the code seems to work well with GPRS, I experienced problems with UMTS.
Here are my test results:
||Device||K850i||W910||W910||K850i||K850i||W910||
|Java Plattform|JP-8|JP-8|JP-8|JP-8|JP-8|JP-8|
|Dataservice|GPRS|GPRS|GPRS|GPRS|UMTS|UMTS|
|Carrier|O2-DE|O2-DE|Vodafone-DE|Vodafone-DE|Vodafone-DE|Vodafone-DE|
|Filesize (Byte)|1.428.077|1.428.077|1.428.077|1.428.077|1.428.078|1.428.079|
|Upload (Byte)|complete|complete|complete|complete|184320 (113664)|627712 (199680)|
The problem of partial upload seems only to exist with larger Files (not sure where exactly the lower bound is).
I hope this was not too much information for a start, but I am currently quite desperate about this bug.
I could really use some help here. I will also provide you with the results of a test on Nokia S60 phones
and the results of SE on device debugging.

The issue was solved by a firmware upgrade [jumpingjoy]
Must have been a problem with JP-8. Works as expected on JP-8.2 now.
||Device|K850i| W910|
||Firmware |R1EA037 |R1EA033|
||Java Plattform |JP-8.2 |JP-8.2|
||Dataservice |UMTS |UMTS|
||Carrier |Vodafone |Vodafone|
||Filesize (Byte) |1.428.077 |1.428.077|
||Upload (Byte) |complete |complete|
HTH
Edited by: er4z0r on 15.04.2008 17:55
Edited by: er4z0r on 15.04.2008 17:56
Edited by: er4z0r on 15.04.2008 17:58

Design a network for voip over wan

Hi all,
I need your valuable suggestions how to design a network of voip using voip.
Suppose there is a Site A and Site B and you are using different service providers and creating different vlans in each site for voice and data.Lets say it like vlan50 for voice and vlan 51 for data and in the other site vlan 100 for voice and vlan101 for data.How do you give the priority for voice packets while sending in the network and basically please explain the needed hardware that you use for this design (like what are the equipments ,media server gateway,dhcp server,ethernet switch , voip phones etc..).
All genius people out there ,kindly help me with this problem.

Hi Rajendra,
Very open query but I will see how much I can give inputs.
Since you have only two site. you can go for CME instead of full CUCM. If you have slightly more sites you can go for CUCUMBE. Or if you have more number of sites you can go for CUCM.
Giving priority to voice packets is given by QOS and the configuration example is given in QOS SRND.
http://www.cisco.com/univercd/cc/td/doc/solution/esm/qossrnd.pdf
Other details will be given in CUCM SRND.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/uc8x.html
You can use sizing tool to see what elase is required.
http://tools.cisco.com/cucst/faces/login.jsp
Regards
Ronak Patel
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