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Working on IP Phone over Broadband Vsat network

sohail_sarwar
Level 1
Level 1

We are using Broadband Vsat Network with an average latency (RTT) around 800ms .

While trying to call from one Vsat Location to other location (Vsat /Radio Network /Leased Line Network) , we are not able to commuincate

Routing is perfectly Okay as we can trace from one router to another and vice versa

What is the minimum threshold of latency inorder IP phones to work

Do we need QOS /traffic shapping etc

Code used is G .729 sampling at 10ms with coding speed of 8 kbits/sec

regards

Sohail

6 Replies 6

kvancoil
Cisco Employee
Cisco Employee

Hi Sohail,

The recommendation G.114 of the International Telecommunication Union (ITU) states that the
one-way delay in a voice network should be less than or equal to 150 milliseconds

This and some more information is available in the SRND:

http://www.cisco.com/en/US/solutions/ns340/ns414/ns742/ns818/landing_uc_mgr.html

Cheers,

Kristof

That recommendation can be safely ignored as anybody in this industry knows.

How remote locations, and ships could have phone service without satellite? Actually they have been doing this since 40+ years.

I personally sometime talk on IP phones over links with one-way delay of 700+ ms and can keep a perfectly normal conversation with even some overlap.

Jonathan Schulenberg
Hall of Fame
Hall of Fame

You may want to check if the satellite equipment is doing optimization on the traffic. Most do and this could be interfering with, corrupting, or even blocking the RTP traffic. I suggest doing a packet capture on either side of the satellite link to see what actually makes it across.

Hi Jonathan

Two days back we were at remote location and after adding IP Phone 7906 to CCM , we have able to make a call from this loction (on Broadband Vsat Site) and one really wonders why it works at some places and fails at most of the sites

Even we are able to dial and listen rining bells but when it comes to commincation , no voice is heard from both sides

DO i need raise the issue with ISP providing BB Vsat connectivity

regards

Sohail

Hi Sohail,

probably you can compare the gateway config between the site, where the call is working and not working.

if you are saying call will ring the phone, but not conversation, it sound like a RTP stream issue or could be

codec issue, which may need dsp resources to do transcoding.

probably you can share your cucm topology with us then we'll have more clearer picture to assist you.

Hi Everyone

Issue still remains unresolved.

We even raised the concern with ISP /Service Provider to check for RTP Stream and what traffic is allowed through their Indoor Units /Vsat Modems .

Just forget to mention in my earlier posts regarding this issue is that for most of the sites we are using GRE Configuration and i hope that has something to do with it. SInce when using GRE , the packets passing through GRE Tunnels are treated as identical

See the URL http://puck.nether.net/pipermail/cisco-voip/2009-June/002111.html

Regards

Sohail Sarwar

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