SIP and VOIP

Hey..
I have almost succesfully published and allowed RTP and SIP through TMG, we can make calls from the VOIP phone, but we are unable to receive calls.
When we call the VOIP phone then it sends UDP traffic ranging from 10.000 to 50.000 and that UDP traffic gets stopped by the TMG. We dont have an internal BPX so we cant publish in that range because we have several phones. What can i do? :)

I got about a similar problem.
Need of internal soft.
phone call to an external PBX
over SIP and RTP protocol ssupport.
I made setting VIOP the wizardin
TMG.We set the Advaced
VIOP setting IP address of thePBX.
When I call the soft.phone
out it's OK.
When someone calls from out side to inside,the phone rings,but
do not hear anything.
Did you have the same problem?
Thank you
RadekRyšan

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X5800 how to use SIP and VOIP

Hello, I have a X5800 music, model 5800d-1 and I can enter all the necessary SIP data. I get the message (settings, connectivity, admin settings, sip settings) registered which is fine. My provider is freephonie. Question: 1) Although on the freebox the SIP is activated, I don't receive incoming call on my nokia. 2) How do I tell my nokia to call on the internet ? Is it possible only using SIP tel numbers which I can enter in the contacts list? Help much appreciated. Pierre.

SIP and VoIP are not synonymous. VoIP can indeed use SIP (and often does), but other services, such as video sharing for example, also use SIP.
It so happens that the 5800 has no native support for VoIP, which is why you can't make SIP calls using it.
Your only solution for SIP is to use third-party services such as fring, for example. Once logged into your fring account you can subscribe to a SIP service. Fring's systems will register to the SIP account and relay the comminucation through your fring account. There's lag and the audio quality isn't as good as a direct SIP connection but at least it works.
The alternative is to use a phone that supports VoIP/SIP natively.
Was this post helpful? If so, please click on the white "Kudos!" star below. Thank you!

Questions about Gateways and VOIP

First~~~ please do not be mad with me.since my questions sound some how stupid.
1. We use gateway between PSTN and IP network. But in PSTN, the voice is analogue. But in IP network, the data is digital. Which role does gateway act? The D/A converter, beside some necessary protocol translation?
2. What is the difference of PSTN and ISDN? I think there is no requirement of gateway between the connection of ISDN and IP network.
3. In PSTN, the signaling protocol is SS7. In IP network, the signaling protocol is TCP for data transmission, and SIP/H.323 for voice transmission such as VOIP. What is the signaling protocol for ISDN?
4. When we talk each other in MSN or ICQ,by microphone and speaker, which protocol in use in this case? still H.323? I do not think so. I guess still TCP?!
thanks for your answer~~~~

Hi lily dont worry we r here to learn!
1. PSTN and IP
A gateway is the point at which a circuit-switched call is encoded and repackaged into IP packets.
For that we use DSP (Digital Signal Processor)
The Gateway converts media provided in one type of network to the format required for another type of network. For example, a Gateway could terminate bearer channels from a switched circuit network (i.e., DS0s) and media streams from a packet network (e.g., RTP streams in an IP network). This gateway may be capable of processing audio, video and T.120 alone or in any combination, and is capable of full duplex media translations. The Gateway may also play audio/video messages and performs other IVR functions, or may perform media conferencing.
In VoIP, the digital signal processor (DSP) segments the voice signal into frames and stores them in voice packets. These voice packets are transported using IP in compliance with one of the specifications for transmitting multimedia (voice, video, fax and data) across a network: H.323 (ITU), MGCP (level 3,Bellcore, Cisco, Nortel), MEGACO/H.GCP (IETF), SIP (IETF), T.38 (ITU), SIGTRAN (IETF), Skinny (Cisco) etc.
I always think in Analog (FXO, FXS) and Digital T1, E1, E&M, etc) and IP packets.
2. PSTN = Public Switch Telephony Network
The worldwide voice telephone network. Once only an analog system, the heart of most telephone networks today is all digital.
ISDN (Integrated Services Digital Network) is an all digital communications line that allows for the transmission of voice, data, video and graphics, at very high speeds, over standard communication lines. ISDN provides a single, common interface with which to access digital communications services that are required by varying devices, while remaining transparent to the user
http://www.ralphb.net/ISDN/defs.html
Yes, you need a GW for the conversion.
3. ISDN for voice transmission Q921/Q931 (Layer 2 and Layer 3)
4. I believe now MSN uses SIP and SIMPLE.
http://www.protocols.com/pbook/VoIPFamily.htm
http://www.packetizer.com/voip/
HTH
//G

Doubt about integration CUCM and VoIP Provider

Hi Guys,
I have the follow doubt: Is possible to do integration CUCM and VoIP Provider using authentication? Is there necessary another equipment to do this? CUBE for example? or another alternative
Thanks,
Wilson

If you are referring to authentication over sip trunk, then NO! CUCM doesn't do authentication you will need a CUBE for that

How to close SIP and H323 port

I have cisco 2801 with IOS (C2801-SPSERVICESK9-M), Version 12.4(5a). I found that my router open TCP port 5060 (SIP) and TCP port 1720. How can I disable this port? I've tried to disable on "voice service voip" but not working. Do you have any suggestion

Hi,
This has been identified as a Bug in CSCsb25337, and was fixed in the following versions..
12.4(7.12), 12.4(7.11)T, 12.4(03d), 12.4(7.14)T, 12.4(07a), 12.4(05b), 12.4(6)T01, 12.4(4)T03, 12.4(2)T05, 12.3(14)YX02, 12.4(2)XB02, 12.4(4)XD02, 12.3(14)YM08
Workaround:
H323 can be disabled by issuing these commands from Config mode:
voice service voip
h323
call service stop
SIP can be disabled by issuing these commands:
sip-ua
no transport udp
no transport tcp
Voice service voip
sip
Call service stop
HTH
Regards
Kevin

How to set up VLAN for DATA and VOIP on SRW248G4P switch?

Hi guys,
I am totally new and was given this task to complete. I really really need help.
We are using one network 192.168.1.0
Shared with data and voip.
CISCO C870, 5 switches LINKSYS SRW248G4P .
The email wrote:-
On the Linksys switch;
- create two different VLANs one for voice and one for data.
- put a firewall between the two VLANs (between voice and data) and only enable certain ports to flow to voice network (inbound tcp 8443 and ssh )
What should i do guys? I really need a dumb guide now.
I know its simple for you guys but i am not a smart IT fella. Whats the step by step?

If the switch is new or you have support on this, then you might try calling the support center. Here is a link:
https://www.myciscocommunity.com/community/smallbizsupport
On the right hand side you can find links to the support center.
Here is a link to the guide:
http://www.cisco.com/en/US/products/ps9967/prod_maintenance_guides_list.html
@ the bottom of this link you can find your switch model, you want the larger of the two. In this guide it shows you how to create a second vlan.
Will your router be the firewall between the two?
Kindest regards,
Andrew Lissitz

SIP and SMTP domains don't match

Hi,
We’ve a situation where SIP, SMTP and AD domain are different than each other and I am sure this will present integration issues at the least...
Speaking from LYNC 2013 and exchange 2010/2013 integration standpoint, could someone please point out to integration process under below circumstances where
SIP domain         uc.contoso.com
SMTP domain      contoso.com
AD domain          contoso.net
Also, how would exactly the logon will occur in this situation?
I'd appreciate if you could point me to relevant document please.
Regards, Pushkal MishrA, This posting is provided “AS IS” with no warranties and confers no rights. I request you to test the solution in lab and post successful outcome you should try this on production.

Hi Pushkal Mishra,
It’s recommend to use the same SIP and SMTP address.
But if they are not the same, they will still work properly if you take some additional work.
I’m assuming that you have published the correct SAN certificate that has every expected SAN name for each domain, then you have to create an additional SRV record for uc.contoso.com.
_autodiscover._tcp.uc.contoso.com
mail.contoso.com        443
In general, Lync will look at your primary SMTP address and attempt to use it to find the EWS URL via Autodiscover after signing in. But if the SIP and SMTP address does not match, you will receive
a security prompt “Lync cannot verify that the server is trusted for your sign-in address”.
To avoid the issue, the best solution in my opinion is to use SRV records for Autodiscover to redirect Lync to the mail.contoso.com A record. (Delete the Autodiscover A record)
Best regards,
Eric

E63 Simultaneous Incoming SIP and Cell Calls

I'm trying to use Google Voice to ring my sip and cell phone at the same time. It seems most of the time only the sip calls ring, and the calls get disconnected as soon as they're picked up. Meanwhile, the cell calls continue as far as the callers are concerned, eventually reaching my mobile voice mail.
There was one time I saw both calls displayed on the screen. I was so shocked that I forgot to see if I could select which call to pick up.

Hi Wendy,
The UC320W is connected to a Sonicwall TZ200 firewall device, which is configured to allow only UDP ports 5060 and 5061 inbound traffic and all outbound traffic. If you can tell me the kinds of things to look for I can check the firewall logs.
Regarding codecs, the SIP provider should be using whatever codec is advertised by the UC320W during registration (or whatever it is that goes on for negotiation). The UC320W's SIP codec settings are set for G711u, so in theory we shouldn't be seeing any G729 in the logs?
Thanks,
Ben

SIP and SCCP

Hi Mates,
I am just wondering if both SIP and SCCP can run on a single router? or should I make one for SIP and another one for SCCP?
Any answers will be appreciated.
Thanks a lot,
Nico

This is Video Over IP, please move to the relevant area.

ASR 1006, are the SIP and SPA cards Hot-swappable?

I can find references that the SPA cards should be hot-swappable in the form of documents stating you should shut down traffic before removing, but I can not find any clear reference that states that the SPA and the corresponding SIP modules can be pulled while the chassis is on.

Both modular SIPs and SPAs are hot-swappable in ASR1K:
http://www.cisco.com/c/en/us/td/docs/interfaces_modules/shared_port_adapters/install_upgrade/ASR1000/asr_sip_spa_hw/ASRmin.html#73220
http://www.cisco.com/c/en/us/td/docs/interfaces_modules/shared_port_adapters/install_upgrade/ASR1000/asr_sip_spa_hw/ASRsin.html#pgfId-1060193
(Text in the second link begins with "Cisco 7600 series router SIPs and SPAs", but that's a typo, the book is about ASR1K (the statement isn't wrong though - SIPs and SPAs are hot-swappable in the 7600 series as well))

Equipments Needed by Newbie to QoS and VoIP

Hello All,
To appease my curiosity on the use of QoS and VoIP, i would like to buy the necessary equipments for my lab experiment but i don't know what to buy.
I currently have routers 2611XM with CSU/DSU, and four serial interfaces each.
What else do i then need to include to allow me practice with QoS, IntServe, Diffserv, VoIP and all that?.
I have read some books mentioned phones or IP phones, would therefore need any help in shaping my ways to the right purchase of equipments.
The sole aim in the long run is building lab for CCVP and hopeful more
Thank you all

You can have a look at this QOS SRND:
http://www.cisco.com/application/pdf/en/us/guest/netsol/ns432/c649/ccmigration_09186a008049b062.pdf

SAP BCM and VoIP technology

Hello Experts,
I just published an article on SDN about SAP BCM and VoIP technology. I believe it can be very helpful for those wishing to learn more about the SAP Contact Center solution and also for those who need a more technical detail about telecommunications.
[SAP BCM and VoIP Technology u2013 A Technical Overview|http://www.sdn.sap.com/irj/scn/index?rid=/library/uuid/30b8eca4-e42d-2d10-668a-fecca15cfbe9]
Hope you enjoy it!
Regards,
Heber Olivar

Hi,
did you find information about the integration of SAP BCM with CUCM? Thanks
Regards,
Alberto

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.
There are also few Mobile service providers who have blocked VoIP on their data network service. So, just check with your service provider. In US both AT&T and TMobile have open network for VoIP.

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 ????????????????

5800 Nokia mobile XpressMusic and SIP for VoIP

Hello;
Anyone can advise for the needed file to be downloaded for the 5800 Nokia mobile XpressMusic to be supported for SIP so I can place voice over IP calls.
Regards
Bilal

Please search this forum for detailed answers to your questions. Summary: there is no Nokia VOIP application for the 5800. You can download Fring and Nimbuzz and connect to your VOIP provider through them.
Lumia 920, Lumia 800
Nokia N8-00 (NAM, Product Code: 059C8T6), Symbian Belle, Type RM-596, 111.030.0609
Nokia 5800 XpressMusic (NAM, Product Code: 0577454) Software v51.2.007, Type RM-428

RV042 protocol binding for SIP and RTP (VoIP)

Hello everybody,
I have a RV042 with a DSL (WAN1) and cable (WAN2) internet connection in Load Balance Mode. The DSL provider also provides internet telephony when registered via his line. When I disable the WAN2 port, my IP phone successully registers with the registration server of the DSL provider. I also defined protocol bindings for SIP (port 5060) and RTP (ports 5004 to 5020) to be bound to WAN1. My IP phone is set up to listen on only these ports.
The rules are in detail:
SIP(UDP/5060~5060) -> "myPhoneIP"~"myPhoneIP" ("RegistrationServerP"~"RegistrationServerIP") WAN1 [Enabled]
SIP(UDP/5060~5060) -> "RegistrationServerIP"~"RegistrationServerIP" ("myPhoneIP"~"myPhoneIP") WAN1 [Enabled]
RTP(UDP/5004~5020) -> "myPhoneIP"~"myPhoneIP" ("RegistrationServerP"~"RegistrationServerIP") WAN1 [Enabled]
RTP(UDP/5004~5020) -> "RegistrationServerIP"~"RegistrationServerIP" ("myPhoneIP"~"myPhoneIP") WAN1 [Enabled]
With these protocol bindings in place when I re-enable WAN2, then after some time the phone reports "registration failed".
Do I need to set something else apart from protocol binding to force the VoIP traffic to go via WAN1?
Thanks for your help
Felix

Pardon my memory if I am mistaken, when configuring the protocol bind for the WAN port, there are 4 or 5 options. Service, which of course is 1~65535, source IP, in this scenario it should be the phone or PBX, whatever you're using. The destination IP should be 0.0.0.0 and interface is your desired WAN, WAN 1 or 2.
Example:
Wan 1- Cable       Wan 2 - Dsl
       |                              |
       | ________________ |
                      |
                  RV042-----------
              ____|                |
              |                     Computer 192.168.10.100
          Tele/PBX 192.168.10.250
On this example to route the Telephone / PBX to WAN 1
All services 1~65535
Source IP 192.168.10.250
Destination IP 0.0.0.0
Interface WAN 1
Please correct me if I am mistaken, I'm currently not at work due to the US holiday

SIP and VOIP

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