E72 SIP to be used for VoIP

Any one can advise me for the needfull to let the E72 able to use SIP service to place voip calls?
Regards
Bilal

Hi BilalGhayad
You need SIP VoIP 3.x Settings (164 kB) from here:http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin...
User Guide available here:http://sw.nokia.com/id/15c504c6-e984-4fd6-be4d-890d24be95c7/SIP_VoIP_Release_3_x_Settings_Applicatio...
Settings application found in Control Panel > Net Settings > Advanced VOIP Settings
Happy to have helped forum in a small way with a Support Ratio = 37.0

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Hi there:
I want to purchase a VOIP Internet phone & line and am concerned that it may not work with my Airport Express. I can use any old phone as long as it is hooked up to the VOIP gateway device which must be connected directly to my High Speed DSL modem. I am currently using the Express and am connected using PPoE. What I need to do is connect the VOIP Gateway directly to my high speed modem and then connect the Express to one of the Gateway's LAN ports.
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Or could l just set Configure to: Using DHCP and it would find the SMC Gateway device since I would be plugged into it's LAN port?
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Thanks so much
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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?
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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.

N8; How to set up SIP and make (WiFI & 3G)VOIP cal...

N8 only comes with SIP software and you need to download the the VoIP setting software at this link (use VoIP 3.1):
http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin...
You can use the N8 web browser and load directly from the site to the phone.
then go to:
Menu>settings>connectivity>Admin.Settings>SIP settings
option>New SIP Profile>use default profile.
This window needs to be configured with your VoIP provider setting which may be found on their site or just do a search. I used the settings from my previous phone. I have Gizmo VoIP and will be using the setting for this service as an example to set up Wifi and 3G SIP setting
Profile Name> Gizmo, (assign the name of your provider e.g.. Gizmo, Google talk, etc)
Service profile> IETF
Default Destination> WiFi
Public username: sip:[email protected] (Input your VoIP provider info)
Use compression> No
Registration> When needed
Use security> No
Then goto Proxy server:
Proxy server address> siproxy01.sipphone.com (use your provider proxy address setting)
Realm> proxy01.sipphone.com (use your provider setting)
Username> (use your username)
Password> (enter your password0
Allow loose routing> Yes
Transport type> UDP
Port> 5060
Go back and then go to Registrar server:
Registrar server address> siproxy01.sipphone.com (use your provider Registrar setting)
Real> proxy01.sipphone.com (use your provider setting)
Username> (use your username)
Password> (enter password)
Transport type> UDP
Port> 5060
Again folks above is just an example. You must use your provider settings.
Then when you are done go back to:
Admin setting> Net setting>Advance VoIP setting>Create new service> choose the SIP profile shown, e.g. Gizmo
This will configure the VoIP setting with the particular SIP profile. It will also add a SIP Tel profile Pane in your Contacts folder, Internet call option to the contacts list and Phone log.
Now go to the SIP pane in Contacts folder, sign in and then make a test call . To make a SIP call just go to a contact and choose Internet call. If you configured everything correctly, it should work and you are done.
For 3G connection, follow above direction and create new SIP profile with new name (e.g. Gizmo (3G). All settings are same except:
Default destination>Internet
Then set up the VoIP setting with the new SIP profile by going to:
Admin setting>Net setting>Advance VoIP settings>Create new service> Choose the SIP profile you just configured e.g. Gizmo (3G)
Then:
Admin setting>Net setting>Advance VoIP settings>VoIP service>choose the 3G SIP profile, e.g. Gizmo (3G)>Profile setting>>>>>
Go down to AWCDMA> set it to "on"
Exit out of everything. Go to SIP (3G) pane in the Contacts folder. Sign in and make test call.
I have configured many Mobile phones and found the N8 somewhat touchy. But, the sound quality on Gizmo VoIP using both WiFi and 3G was great, no echo or any problems.
Useful hints:
If it does not connect or make a call, you need to check the SIP setting for any error. If you find any problem there, you must first remove the SIP profile in the Net setting. Then go to the SIP setting and correct the problem. Double check everyting, go back to Net setting>Advance VoIP setting>Create new service and add the SIP profile. The reason is that when you Configure a VoIP setting with a SIP profile, the SIP info is permanently loaded and will not change even if you make any change in the SIP profile. So any time you change anything in the SIP profile (even the password) you always need to remove, make correction and then reload the corrected SIP profile to the VoIP setting.
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Hi
I have an unbranded N8 and am trying to get VoipCheap to work.
I have tried to set up VOIP but still facing a problem or 2 so please assist me.
I have an active account via VoipCheap.com with a profile name and password and it is verified.
1/ When I look in Admin Settings / Sip Settings it shows VoipCheap with "not registered" under it..why?
My settings when I click on VoipCheap are:
profile name: VoipCheap
Service profile: IETF
Default destination: Internet
Public username: sip:[email protected]
Use compression: No
Registration: When needed
Use security: No
Proxy server address: sip:sip.VoipCheap.com
Realm: VoipCheap.com
Username: [email protected]
Password: ****** (I have entered my password)
Allow loose routing: Yes
Transport type: UDP
Port: 5060
Registrar server settings same as Proxy server settings.
I have also gone to Net settings / VoipCheap settings and see:
Username: sip:[email protected]
Password: ***** (I have entered my password)
Default service: No
Service connectivity: Internet
In Contacts I have a VoipCheap tab with the icon for internet calling and it says
Sign in (example)
but when I click on this it shows connecting (via my internet connection) but hangs there - it does not seem to connect............
what do you think I am doing wrong, please ????????????????

Is QoS tools shape good for VoIP?

Hi,
I have MPLS VPN scenario. For MPLS VPN at PE router connecting to hub, I've configured below QoS:
class-map match-any P1
match ip dscp ef
class-map match-any P3
match ip dscp default
class-map match-any P2
match ip dscp af31
policy-map QoS
class P1
priority percent 35
class P2
bandwidth percent 40
random-detect
class P3
bandwidth percent 25
random-detect
Since on the egree PE towords the internet, I'm using subinterface thus, the configuration is:
policy-map cust-qos
class class-default
shape average 256000
service-policy output QoS
interface FastEthernet1/0.8
encapsulation dot1q 8
ip address ...
service-policy output cust-qos
Since the internet access is 256K, I've configured shape average 256000.
My problem is that, during high congestion at internet access, I noticed that on cust-hub my VoIP SIP is having problem to ring to another VoIP SIP within the same hub. Is it because of the shape command ?
thanks in advanced

Hi Maher,
shape average 256000 is to much. The point is, that you have OSI layer2 overhead and layer2 keepalives and such. Lower it to f.e. 240000. You might need to "play" a little with the parameter, as the overhaed depends on average packet size.
Also make sure, that your SIP traffic is marked with af31. Just as a note: Cisco changed from marking voip signaling with af31 to cs3. Take a sniffer and check that proper markings are in place for voip signaling. Or just try
class-map match-any P2
match ip dscp af31 cs3
Hope this helps! Please rate all posts.
Regards, Martin

External Ringer for VOIP Phone

I have external ringers connected to VG224's for VOIP phones in our shops. I have them configured and they work as they should for a while, but after a month or two it appears to blow up the port on the VG it is attached to. I am thinking it might have to do with resistance of the bells, but since I am not an electrician I am not sure. The ringer is a Premier PT-102 bell and per the description is 1000/2650 Ohm. Has anybody experienced problem with external ringers blowing up VG ports. Or is there any alternatives to the analog ringers that is designed to work with the Cisco VOIP phones. Thanks for any info anybody can supply.

I did get one of the Algo 8180 SIP Audio Alerters and it works great as an external ringer. Also since it has the capability of paging, we are trying to get it configured to use with our IPCelerate paging solution. Does anybody have any thoughts or ideas how to accomplish this. I can get it to page using a direct extension to the Algo device, but can not get it to receive pages from IPCelerate. Any help would be appreciated.

ISA550 DMZ for VOIP

Hi !
I have a ISA550 and I want to make a DMZ LAN with VOIP SIP Phones ! This DMZ Subnet schould not be patet and should ve accessed from the Internet only form a certian Subnet !
The DMZ Network is statc routed to the ISA 550 WAN Interface from outside !
Here my Config:
ACL Rule
From WAN to DMZ Services any Source Managment Destination Address DMZ-Network enable
Dynamic PAT
disable DMZ
Advanced NAT:
From DMZ to WAN1 Original DMZ to WAN1 Orgnal Source DMZ-Network Orginal Destination any any Translatated Source DMZ-Network any any
The config above not work ! What is wong ?
Also I want to use QOS with strict priority Queuing for this Network -> How should I do this -> Protocol RTP and SIP
Thanks !
kind regards
kristoferus

Kristoferus,
To accomplish what you are wanting shouldn't require any special ACL Rules or NAT configuration. The reason for this is because all communications are initiated from the SIP Phones to the VoIP provider. Once the SIP Phone connects to the VoIP provider's network, all traffic is managed over that established connection. So the VoIP provider's network will never actually initiate a connection into your network and thus you don't need any special NAT configuration to allow them access.
Since the VoIP provider is in a lower security level Zone (WAN), the VoIP DMZ will have, by default, unlimited access to the WAN, and by default the WAN will explicitly allow back through any traffic initiated from a higher security level Zone (VoIP DMZ). So this is what I would suggest doing.
Assumptions
You mentioned source Management in your ACL Rule. I'm assuming Management is the Address Group that contains the VoIP provider's subnet.
Steps to complete
Delete the ACL Rule you created and referenced above
Delete the Advanced NAT statement you created and mentioned above
Under Dynamic PAT, enable for the VoIP DMZ as well
Under Application Level Gateway, ensure SIP Support is unchecked
Under Networking -> Ports -> Physical Interface, ensure the DMZ VoIP VLAN is applied to the appropriate interface
If the only thing attached to this port is an external switch that only phones plug into, then ensure the mode is set to Access and that the only VLAN on this port is the VoIP DMZ.
If a switch is connected to this port and that switch is shared by both phones and devices in other VLANs...
Ensure the external switch supports VLANs
Set the Mode to Trunk and add the appropriate VLANs to the port
Configure the external switch ports to include the correct VLANs on the associated ports to ensure that phones are being placed in the VoIP DMZ VLAN and not the default.
Once you have all of this setup, test the phones and they should be working. If they are proceed to the next step to lock down security. If not, do not proceed to the next step as it will only add complexity to troubleshooting.
Limit SIP Phones to only have access to VoIP provider
Create an ACL Rule...
From Zone: VoIP DMZ
To Zone: WAN
Services: ANY
Source Address: VoIP DMZ Network
Destination Address: Management (Based on my assumption above)
Log: On
Match Action: Permit
Create an ACL Rule...
From Zone: VoIP DMZ
To Zone: WAN
Services: ANY
Source Address: ANY
Destination Address: ANY
Log: On
Match Action: Deny
Once both ACL Rules are created, ensure the first ACL Rule is before the second ACL Rule.
The combination of these rules in the correct order will allow traffic from the SIP Phones to the VoIP provider and then block the SIP Phones from accessing anything else.
You should be done and working at this point. If so, let me know and we can move on to QoS.

E70 - conecting with WLAN for voip/internet calls

Hello,
I have nokia e70 and i'm trying to connect to sip proxy server but i can't
The reason is that i cannot connect to WLAN by making voip/internet call.
I can connect to WLAN when I run browser or getting emails.
Is there way to connect my phone to WLAN without running browser or email client, and make e70 used this connection for voip calls ?
maybe this is a bug, my software version is :
1.0610.05.07
30-05-06
RM-10
Did someone successfully get voip calls with WLAN on this software version ?
Please HELP !
Best Regards
Slawomir Swider

First make sure your 'Default access point' is configured under your SIP profile. Second, make sure your Intenet Tel. Profile is assigned to using your SIP profile.
You can manually register your SIP service by going to 'Menu -> Connectivity -> Internet Tel. -> Registration Status' and change its status. It will connect to your WLAN automatically and register. It can take a whole minute depending on your connection, so be patient.
You don't need to manually register when just making calls; punch in the desired number then 'Option -> Call -> Internet Call.' It'll prompt you to 'Register now' if you haven't.

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
Please rate helpful posts by clicking stars below the answer.

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Hey Anthony,
Yes, a lot to read, thanks for reading though!
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E72 SIP to be used for VoIP

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