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Author: Gilson

why one-way audio problem?

why one-way audio problem?

Almost all of us will meet this problem when we deploy our first VoIP network. We are often confused: why I cannot hear peer guy but he can hear me? why we cannot hear each other?

The root reason is that there is private network and public network. If both sides are in different network, the problem will happen. Please look at below figure which desribe a very simple VoIP network:

One way audio problem network topology

In this simple network, we have two VoIP devices, one is SIP phone whose number is 100, another is SIP client whose number is 101.

SIP phone is in a private network and its private address is 192.168.1.100, and its router is connected to public network with address 8.8.8.8.

SIP client is installed in one PC which is in the public network with address 8.8.4.4.

So when SIP phone makes a call to the SIP client, what will happen?

SIP phone say: Hi, I am 100, my audio address is 192.168.1.100. Please send audio stream to me.

SIP client answers it: ok. I am 101, my audio address is 8.8.4.4. Please send your audio to me.

SIP phone sends audio stream to SIP client. Since “8.8.4.4 ” is a public address, it is no problem for SIP client to receive the audio stream from SIP phone. That means SIP client can hear SIP phone now.

SIP client sends its audio stream to SIP phone “192.168.1.100”. You can see it is a private address and cannot be reached by SIP client which is in public address. SIP client will fail to send its audio stream to SIP phone in fact.

So finally, SIP client can hear SIP phone, but SIP phone cannot hear SIP client. This is a very typical one-way audio problem.

Then, how can we resolve it? To be continue …… 🙂

Can miniSIPServer support G.729, iLBC, GSM audio codec, or video calls, and so on?

Can miniSIPServer support G.729, iLBC, GSM audio codec, or video calls, and so on?

We are often asked this question about audio codec or video calls. Some customers want to confirm whether miniSIPServer can work well with their SIP phones/clients who can support several audio codecs or video calls.

We often answer that it depends on their own SIP phones/clients. miniSIPServer, not matter local-miniSIPServer or cloud-miniSIPServer, doesn’t care media codec by default.

Why we say that?

Please have a look at following figure which describes a basic media process model of miniSIPServer in a normal basic call.

From this figure, we can see:

(1) miniSIPServer only controls call signals.

(2) Media stream is processed by the clients themselves.

This model will reduce the work-load of the server since all media streams are bypass server, so it is unnecessary for the server to support several media codecs.

But in some service scenarios, miniSIPServer need play audio to the SIP phones. For example, in auto-attendant or calling card services, miniSIPServer need prompt audio announcement and collect user input information. Obviously, miniSIPServer is required to support audio codecs in these scenarios.

That’s right. If miniSIPServer need process audio stream, it can support two audio codecs: G711a (PCMA) and G711u (PCMU).

By the way, after playing audio stream, the final media stream will be processed still in end-to-end model. Please refer to following media model of miniSIPServer for these scenarios:

Play audio stream
Why I cannot receive voice mail?

Why I cannot receive voice mail?

When some customers use miniSipServer cloud ( for local minisipserver, the same) to process their voice mail, but they are often confused why cannot receive voice mail in their email box.

In fact, we have an online document to describe how to use voice mail feature. Please refer to:

http://www.myvoipapp.com/docs/mss_services/voice_mail/index.html

For MSS cloud, there is a little difference. First, MSS cloud doesn’t support MWI (message waiting indication), so it can only send voice messages to your email address. Sometimes, customers often forget to configure SMTP server information which is necessary for MSS cloud to send your voice mail, or they often forget to cofigure email address of specific extensions.

So let’s describe more details about these. Before this, I assume that you have signin into your MSS cloud account. If not, please sign-up one.

Step 1: configure SMTP server information

smtp configuration in mss cloud system

Please look at the figure. You need click “system information” linker, then fill your SMTP information there. By the way, for Gmail accounts, they are always required to configure secure connection.

If SMTP information are all right, MSS will use it to send voice mail.

Step 2: configure email address of extension

configure email address of extension

Please click “local users” linker and select or add one extension to edit. For voice mail feature, you must configure “eMail address” to make MSS cloud know where the voice mail should be sent to.

Of course, you also need configure “voice mail” service right to this local user / extension.

So after that, for no-answer incoming calls, MSS will prompt the caller party to leave messages and save/mail them to the user’s email address.

miniSipServer cloud released

miniSipServer cloud released

We are very glad to announce that miniSipServer cloud is released today.

“miniSipServer cloud” is one kind of host-pbx service for small business and customers can add more or reduce communication capabilities according to their real requirements. It is very flexible.

Customers will not need to setup local sip server or pbx. It is in cloud and it is FREE to try as many times as you want.

“miniSipServer cloud” is built with miniSipServer and Amazon EC2.  It is the beginning of our cloud-communication. It is so great and we decide to assign “minisipserver.com” to it. so you can begin to try the latest cloud-pbx at following linker:

http://minisipserver.com

V3 upgraded to V3.1.5

V3 upgraded to V3.1.5

By default, there isn’t administrator password in MSS. So if the customers enable HTTP server, it could be very dangerous without password protection.

In this new upgration, the HTTP server will not be startted if there isn’t administrator password in system configuration.

This modification is also merged to latest V4 version.

LTS, stable, development versions upgrated

LTS, stable, development versions upgrated

LTS version is upgraded to V2.10.7 to discard 3PCC process in pickup service because some old voip gateways or phones cannot accept 3PCC process.

V3 is upgraded to V3.1.4, and V4 versions are upgraded together to include above modification. In additional, they are also updated to fix some bugs in web management system.

what will next version be?

what will next version be?

We are working on pushing miniSipServer to be a cloud-pbx system.

What’s that mean? It means it is unnecessary for our customers to maintain a PC or server to run miniSipServer. We can use a miniSipServer in cloud as our own ‘virtual’ PBX, and it will be very easy and flexible to add or reduce communication capabilities. For example, the current miniSipServer is separated to 20/50/100/300/1000 clients version, but for some small business customers, maybe they need 5 local users and hope to add more with the growing of their business. It will be available in cloud-pbx system.

Most features will be same with current miniSipServer running on PC/Server, except that it is running in cloud. It will be same for customers to maintain the cloud pbx through web management system.

45meeting lauched

45meeting lauched

45meeting website is lauched today!

This is a website to provide simple online conference service for small business. It is very easy to use and interesting. And we hope it can benefit our customers.

Please enjoy it. The website is http://45meeting.com.

how to link fxo gateway?

how to link fxo gateway?

It is a often asked question. In MSS, you can use “external line” to connect to your fxo gateway. In our forum, we give a simple description about how to link to linksys 3102. Maybe you can refer to this document for help.

http://www.myvoipapp.com/forums/viewtopic.php?f=5&t=419

(Sorry, since our forums have been removed, above document cannot be visited. Please refer to online F.A.Q document. Thanks. )

miniSipServer LTS upgrade to V2.10.6

miniSipServer LTS upgrade to V2.10.6

MSS LTS is upgraded to fix following problem or provide new feature:

(1) Fix a bug in GSM audo codec.

(2) Limit the max simultaneous call of FXO external line to only 1 by default since most FXO gateway can only support 1 call for one line.