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Remote Softphones


TimB
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I have a client thats opening a satellite office and wants to use 3 soft phones in that office. I looked in the manuals and don't any solid instructions on how to setup up remote soft phones. Client is running ver. 1.5.1.1a in the main office with a Time Warner Cable modem with a dynamic IP. The remote office will also be a Time Warner Cable modem with a dynamic IP. Any suggestions would be appreciated.

 

Many Thanks

Tim

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Well, first of all check out http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses and http://wiki.pbxnsip.com/index.php/Troubles..._Trunk_Problems.

 

Also, from our experience dynamic IP addresses will cause a lot of random instability and problem searching. When the IP address changes you will have "blind spots" where the PBX is not connected to the remote phones. Also, we learned that our ISP likes to do service windows at times when we didn't expect it. Needless to say, that if the ISP does not support QoS routing and/or does not use a session border controller, you will sooner or later end up with choppy audio. Our sad experience is that when an ISP also provides (TDM-based) telephony services, they have zero interest to respect QoS. Just setting expectation levels - you get what you pay for.

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Well, first of all check out http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses and http://wiki.pbxnsip.com/index.php/Troubles..._Trunk_Problems.

 

Also, from our experience dynamic IP addresses will cause a lot of random instability and problem searching. When the IP address changes you will have "blind spots" where the PBX is not connected to the remote phones. Also, we learned that our ISP likes to do service windows at times when we didn't expect it. Needless to say, that if the ISP does not support QoS routing and/or does not use a session border controller, you will sooner or later end up with choppy audio. Our sad experience is that when an ISP also provides (TDM-based) telephony services, they have zero interest to respect QoS. Just setting expectation levels - you get what you pay for.

 

 

I understand what your saying becouse the IP can change on a regular basis. There should still be away to make this work without changing to a static IP or getting 2 cable modems. This remote office won't be used on a daily basis so if the IP address changes back at the main office they can just make the config changes on their softphones.

I have done a handful of Asterisk installs with the same setup with a dynamic IP on the cable modem and opening ports on the router to allow SIP traffic to flow in and out. Is this just not possible with pbxnsip??

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I understand what your saying becouse the IP can change on a regular basis. There should still be away to make this work without changing to a static IP or getting 2 cable modems. This remote office won't be used on a daily basis so if the IP address changes back at the main office they can just make the config changes on their softphones.

I have done a handful of Asterisk installs with the same setup with a dynamic IP on the cable modem and opening ports on the router to allow SIP traffic to flow in and out. Is this just not possible with pbxnsip??

 

 

OK..forwarding port 5060 in the router to the pbxnsip servers lan address of 192.168.1.141 allows me to registers and xlite phone using the cable modems wan ip and I can dial other extensions of the system. I have no audio yet but working on it.

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OK..forwarding port 5060 in the router to the pbxnsip servers lan address of 192.168.1.141 allows me to registers and xlite phone using the cable modems wan ip and I can dial other extensions of the system. I have no audio yet but working on it.

SIP is, generally speaking, SIP. If Asterisk can make it work so can PBXnSIP. All of the points that pbxnsip (the user) brought up here are valid. If it doesn't support QoS you will eventually have issues. Please consider this carefully and plan for it when you dimension and deploy your PBXnSIP servers.

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SIP is, generally speaking, SIP. If Asterisk can make it work so can PBXnSIP. All of the points that pbxnsip (the user) brought up here are valid. If it doesn't support QoS you will eventually have issues. Please consider this carefully and plan for it when you dimension and deploy your PBXnSIP servers.

 

 

I now have a Snom 300 registered as a remote extension but I still have no audio. I can call out to the pstn but when I establish a connection theres no audio path, same when calling another extension. I have opened up ports 49152-64512 in the router. Any suggestion would be welcomed.

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"Also, from our experience dynamic IP addresses will cause a lot of random instability and problem searching."

 

SIP is not HTTP...

 

 

 

FYI-We have it working fine now. Its nice to have a static but we have figured out you can make this work with a dynamic IP with pbxnsip NATED by port forwarding in your router. With good bandwith 3 remote phones should be fine QOS wise.

 

time warner dynamic IP cable modem> linksys router>pbxnsip on the lan.

 

remote phone registers with the dynamic wan IP address for the time warner cable modem and registers correctly.

 

If the wan IP changes you just change it in the phones GUI.

 

You could also use a free dns service like dyndns.com and give your server a name voip.whatever.com and register your remote phones that way too.

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  • 3 weeks later...
WOW. Respect!

 

 

Well I thought I had this figured out, NOT! The remote extension can dial another extension back at the main office and it works fine, they can also grab dial tone from the main office and use the trunks for out bound calls. But when an extension at the office calls a remote extension I can hear them but they cannot hear me. I have played around with firewall ports on the remote end routers with no results.

 

Any suggestions?

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Without a SIP trace/ethereal dump it's very hard to see what's going on. Can you maybe run Ethereal on the PBX and on the softphone PC, configure them both to capture and initiate the (failing) call. The one way audio sounds like the softphone is replying to the INVITE saying "send you audio here", the PBX then sends the audio to a) The wrong IP B) The incorrect port or c) The router has restrictive NAT policies. STUN should discover the type of NAT for you.

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Without a SIP trace/ethereal dump it's very hard to see what's going on. Can you maybe run Ethereal on the PBX and on the softphone PC, configure them both to capture and initiate the (failing) call. The one way audio sounds like the softphone is replying to the INVITE saying "send you audio here", the PBX then sends the audio to a) The wrong IP B) The incorrect port or c) The router has restrictive NAT policies. STUN should discover the type of NAT for you.

 

I will run the SIP trace, going back onsite today. Can you recommend a STUN server? Its not a softphone but a Snom 300 at the remote site. It just baffles me that the remote site can call the main office and it works fine but when the main office calls the remote site there's only one way audio.

Would a VPN between locations solve my problems?

Thanks

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stun.xten.com I think should work ok.

 

I think its because the Snom 300 initiates the call and opens up the necessary pin-holes in the router/firewall - so symmetric RTP is working. When the remote office initiates the call, the sessions dont exist so are dropped. It's very hard to say though - it varies with different models of router and UA.

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stun.xten.com I think should work ok.

 

I think its because the Snom 300 initiates the call and opens up the necessary pin-holes in the router/firewall - so symmetric RTP is working. When the remote office initiates the call, the sessions dont exist so are dropped. It's very hard to say though - it varies with different models of router and UA.

 

When the remote office initiates the call it works great. Its when the main office initiates the call to the remote office theres a problem.

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Guys, don't waste your time on STUN. If you send traffic to the PBX then there is not need for STUN. If a phone is behind NAT, it should tell the PBX by presenting a private IP address then the PBX can take care about it (if it is possible at all).

 

STUN for SIP was a mistake, it only works in the scope of ICE.

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