Remote Phone

[nixter80]

Hi, I have a CCM running at a location.

I have a need to have a remote phone connected to the server via a satelite link, which is high latency.

I can make and receive calls sucessfully across the link, the only problem is that on the remote phone, I can hear the other person, but they can not hear me.

I am using g729.

If i plug the phone into the local lan, it works fine.



I think it might be because hte packets are timing out because of the hight latency involved - pings aroung 800ms.


Any ideas how i can increase the timeout values, or if it might be onother problem.

 

[nixter80]

Phone is a 12sp+. ping times across satelite approx 800ms

 

[computerhighguy]

This absolutley will not work. VoIP needs 150ms or less round trip time to ensure good quality voice and that could be extended to 200 MS as the MAX. 800ms is not an option here. I had a guy on satellite that his average ping time was 2 seconds +. Satellite is good for 2 things. Web surfing and non time sensative batch file transfers. You will need to come up with another solution. It realyy doesn't matter which vendor you go with, 800ms+ is too much time.

 

[Conleyjo]

This absolutely will work because I am doing it! It doesnt matter how long the latency is as long as the packets arrive in the correct order. The conversation might sound like a CB radio depending on the delay, but it will work, and it is a free call. I just talked to a friend in Kosovo this morning who is on the other end of a VPN tunnel over a Sat link. Most of the time the call quality is soo good, people dont know he is 7000 miles away.

Here is the key to making this work. You must assign the phone on the remote end with a routable IP address that is internal to your network. Do this by establishing a VPN tunnel at the remote site, then give the IP phone an internal routable address via DHCP or give the phone a fixed IP address.

Hope this helps.
John

 

[commsguy]

ConleyJo is right, latency is only rlevant for quality issues, not functionality. I have several clients using Voice over Satellite IP. They use it much like a walkie talkie (i.e. hello, over)

The issue here is a one way voice problem.....period. You may have to adjust timeouts but it should work.

You need to provide more information as to what happens on the call....does it ring busy after a few moments of silence?

Please post detailed call progression here.

Thanks,



commsguy

http://www.convetech.com

 

[nixter80]

Ok, This is the set up I have.

At the main site
Cable connection to the internet with a vpn server.
Behind the vpn server I have a cisco callmanger, a bunch of ip phones, a dns server, and a h323 gateway to the public phone network.


At the remote site I have a vpn server.
Behind the VPN server is an ip phone.



The two vpn servers connect to each other and bridge the two network 192.168.5.0 at the main site and 192.168.8.0 at the remote site.



The remote ip phone registers sucessfully with the call manager. It is a 12sp+ set up to use g723.

I have placed a packet capture program on the ip phone and the h323 gateway.


What happens is as follows.

I pick up the handset of the remote ip phone.
after about .5 seconds the phone gets dialtone.

I dial another extension at the main site or a number that will use the h323 gateway.

After about .5 seconds that phone starts ringing.

When i pick up the phone at the main site, anything i say comes out at the remote site fine, with about a .5 second delay.


When i speak into the phone at the remote site, Only very small bits of what i say come at at the main site, and very choppy.


I tested but putting the remote site onto another cable connection, and set a vpn up that way, and it worked fine, which is why i belive that it is a problem with the ttl values of the packets.


The the packet capture at the main site i see very few packets from the remote site, but at the remote site i see heaps of packets being sent, so I guess it is a router or something along the way, that is dropping the packets.

Thanks,
Nichol

 

[rleestma]

I have one way voice problems too... What i say at the remote site works fine, but what I say from the hq site TO that phone isn't heard at all...

Ideas??

Ryan

 

[nixter80]

Ryan,

What caused the problem for me was the method used to send the data via sattlelite. Downstream worked fine, because it was a brodcast channell, so the packets got through pretty fast.

Upstream, because you could only transmit for a small portion of time, the packets were stored, then all transfered in one block. This caused problems because the voip packets were timing out.

Thanks.

 

[Wanabeelee]

I also have a one way voice problem but for a diff reason.

I am useing routing to let a 7912 that is behind a linksys router connect to a callmanager that is behind a firewall over the internet.

Yes I know, this should not work. I have got the phone to connect to callmanager and I can dail and I get calls, but no voice ever reaches the remote phone. In callmanager it is regeistered as the local ip of the phone behind the linksys. No where for the return voice to go is my guess cause that ip is not on the callmanager network.

So check your callmanager to see what the other phone is regeistering as and that might help you find the one way voice problem.

And if anyone can shed light on my problem *smile*

 

[rleestma]

We need to wait for the next version of CME... Then it will be supported