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Audio problem SIP phone
- Thread starter elperkins
- Start date
When I first got started and was able to get my phone to connect and stuff but having audio problems I found that my codec preferences did not match.
I also found that when using the Cisco 7940 SIP phone that I had to have the incoming and outgoing proxy set to my gateway IP (your HG1500 IP) and I had to have a checkmark in the enable proxy (or something like that) check box.
I would also throw my 2 cents worth in and say that if you have the resources to do it you are much safer keeping the IP of your HG1500 behind the firewall and setting up a VPN connection into your network than you are pointing the DMZ to it and opening it wide up to anything that might come in from outside, but that's just me being paranoid!
I also found that when using the Cisco 7940 SIP phone that I had to have the incoming and outgoing proxy set to my gateway IP (your HG1500 IP) and I had to have a checkmark in the enable proxy (or something like that) check box.
I would also throw my 2 cents worth in and say that if you have the resources to do it you are much safer keeping the IP of your HG1500 behind the firewall and setting up a VPN connection into your network than you are pointing the DMZ to it and opening it wide up to anything that might come in from outside, but that's just me being paranoid!
- Thread starter
- #3
Hi, thanks for your reply.
I know that setting up a VPN connection is the best option, but not for my customer, as he has IP phones (SIP) around the world, wich are mobile, moving from one place to another.
Is there any possibility of having audio in both ways without setting a VPN connection?
So why doesn`t it work if I open all the ports in spite of security risk?
I have audio in both ways only if I set the public IP directly in LAN1 interface of HG1500 card.
Another problem I have is the following: when I make calls through my SIP Provider I have audio in both ways, but when I receive a call from PSTN through my SIP Provider it´s me the only one with audio.
Thanks!
I know that setting up a VPN connection is the best option, but not for my customer, as he has IP phones (SIP) around the world, wich are mobile, moving from one place to another.
Is there any possibility of having audio in both ways without setting a VPN connection?
So why doesn`t it work if I open all the ports in spite of security risk?
I have audio in both ways only if I set the public IP directly in LAN1 interface of HG1500 card.
Another problem I have is the following: when I make calls through my SIP Provider I have audio in both ways, but when I receive a call from PSTN through my SIP Provider it´s me the only one with audio.
Thanks!
I don't know anything at all about SIP trunking, but if the calls are working properly from your SIP provider and not from the PSTN I would look at your PSTN gateway configuration.
As for the phone, it should work fine with the DMZ host wide open - that's why I thought maybe the codec preferences were messed up or the proxy/voip gateway settings, or it could be something in the station configuration for that extension.
I don't understand the issue with VPN and moving around. I have a VPN connection to my employer's network and it connects wherever I am - I just tell it to connect, enter a few logins and passwords and I'm good to go. Doesn't matter where I am as long as I have an internet connection!
As for the phone, it should work fine with the DMZ host wide open - that's why I thought maybe the codec preferences were messed up or the proxy/voip gateway settings, or it could be something in the station configuration for that extension.
I don't understand the issue with VPN and moving around. I have a VPN connection to my employer's network and it connects wherever I am - I just tell it to connect, enter a few logins and passwords and I'm good to go. Doesn't matter where I am as long as I have an internet connection!
Iamnothere
Technical User
Most likely a CODEC problem. Compare the phone's CODECS to those enabled on the HG. There must be at least one match!
- Thread starter
- #6
You need a STUN server, the siemens will send the internal ip addres 012345679@192.168.1.1 to the public server. Using STUN will solve this 012345679@89.1.1.2 so you RTP stream won't come up. You could use any free STUN server for this.
I mostly use stunserver.org
Public STUN servers
•stun.ekiga.net (alias for stun01.sipphone.com)
•stun.fwdnet.net (no XOR_MAPPED_ADDRESS support)
•stun.ideasip.com (no XOR_MAPPED_ADDRESS support)
•stun01.sipphone.com
•stun.softjoys.com (no DNS SRV record) (no XOR_MAPPED_ADDRESS support)
•stun.voipbuster.com (no DNS SRV record) (no XOR_MAPPED_ADDRESS support)
•stun.voxgratia.org (no DNS SRV record) (no XOR_MAPPED_ADDRESS support)
•stun.xten.com
•stunserver.org see their usage policy
•stun.sipgate.net:10000
•numb.viagenie.ca ( (XOR_MAPPED_ADDRESS only with rfc3489bis magic number in transaction ID)
•stun.ipshka.com inside UA-IX zone russsian explanation at
Also check to wich ports the provider send the RTP stream, mostly they use 15000-53000 UDP. So those port also need to be open.
So only FWD 5060 is not enough.
Don't know if you have the option in the siemens to say "use provider codec" or something like that.
___________________________________________
It works! Now if only I could remember what I did...
Dain Bramaged
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I mostly use stunserver.org
Public STUN servers
•stun.ekiga.net (alias for stun01.sipphone.com)
•stun.fwdnet.net (no XOR_MAPPED_ADDRESS support)
•stun.ideasip.com (no XOR_MAPPED_ADDRESS support)
•stun01.sipphone.com
•stun.softjoys.com (no DNS SRV record) (no XOR_MAPPED_ADDRESS support)
•stun.voipbuster.com (no DNS SRV record) (no XOR_MAPPED_ADDRESS support)
•stun.voxgratia.org (no DNS SRV record) (no XOR_MAPPED_ADDRESS support)
•stun.xten.com
•stunserver.org see their usage policy
•stun.sipgate.net:10000
•numb.viagenie.ca ( (XOR_MAPPED_ADDRESS only with rfc3489bis magic number in transaction ID)
•stun.ipshka.com inside UA-IX zone russsian explanation at
Also check to wich ports the provider send the RTP stream, mostly they use 15000-53000 UDP. So those port also need to be open.
So only FWD 5060 is not enough.
Don't know if you have the option in the siemens to say "use provider codec" or something like that.
___________________________________________
It works! Now if only I could remember what I did...
Dain Bramaged
___________________________________________
- Thread starter
- #9
I don't know if that's the problem. My SIP phones work just fine as long as I can get to the gateway IP on the 4000 - I don't have any weird configuration set up.
Look in the Basic3 tab at the Codec Type. Note if you have chosen ALWAYS to be used or PREFERABLY to be used. Don't choose Always - that gives more flexibility in the codec selection.
This shouldn't be goofed up, but go into Bus Extension tab and look at Protocol. SOPP/S2PP should be set to SBDSS1.
There was one other spot where the Codec setting applies but I couldn't find it quickly and it's Monday morning - my phone is ringing off the hook. Make sure no other codec settings in the phone or the 4000 are set to Always use only one protocol - just pick the preferred one.
The other setting is to choose compressing codecs vs non-compressing codecs and that needs to be set the same too.
Look in the Basic3 tab at the Codec Type. Note if you have chosen ALWAYS to be used or PREFERABLY to be used. Don't choose Always - that gives more flexibility in the codec selection.
This shouldn't be goofed up, but go into Bus Extension tab and look at Protocol. SOPP/S2PP should be set to SBDSS1.
There was one other spot where the Codec setting applies but I couldn't find it quickly and it's Monday morning - my phone is ringing off the hook. Make sure no other codec settings in the phone or the 4000 are set to Always use only one protocol - just pick the preferred one.
The other setting is to choose compressing codecs vs non-compressing codecs and that needs to be set the same too.
I know the HiPath doesn't support STUN in softphones, try 3CX, sjphone or X-lite to test, enter a free stun server and see if that works. If that works and the Siemens hard/soft phone doesn't then you need a different setup on the HiPath. You then could try to set the router in bridge mode so your Hipath gets the public IP address, if you can't then you'll never get it to work.
___________________________________________
It works! Now if only I could remember what I did...
Dain Bramaged
___________________________________________
___________________________________________
It works! Now if only I could remember what I did...
Dain Bramaged
___________________________________________
Iamnothere
Technical User
Are you a Siemens instructor? If so, where?
- Thread starter
- #13
- Thread starter
- #14
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