I really curious about the quality of VOIP over internet. Is there any possiblity to do this by using cisco router at both end like below?? If possible, how can it be implemented?
(site A)IP-Phone--------(Cisco router)---------INTERNET_CLOUD----------------(Cisco router)-------------IP-Phone(Site B)
Really appreciate your advice.
If you create a VPN tunnel between your routers you can put voice traffic across.
Just be aware that there is no QOS through the internet, so no guarantee of quality or reliability. Many people do this though and it just depends on your particular needs. Skype has hundreds of millions of users and operates over the internet. For business customers, I generally consider that customer facing voice traffic over the internet is usually not appropriate, but inter-office voice traffic is generally fine because the expectation of quality and reliability can be lower for inter-office and the potential cost savings can outweigh the poor quality and/or reliability.
Your model will work as we have tested this hundreds of times however as the other poster noted there will be times when, because of no QoS, there will be quality issues. Regarding the VPN, be sure to only tunnel the signaling and not the RTP. If you encrypt both the signaling (SIP/MGCP/H323/...) and the audio RTP stream it is likely you will create jitter and audio problems.
Really happy when I read your comment regarding this model setup. Do you have any reference on how to configure this kind of model and make it works without lagging the voice performance and quality?? Really appreciate your advice.
What do you have available, and what are you using. Are you trying to do a Cisco IP phone with a Call Manager and SCCP, or an Aastra phone with an Asterisk PBX... The key is the PBX/Manager/Switch and how you program it to send the RTP resource for the audio stream.
Yiu can find all kinds of documents usign the CCO serach box.
However, since it appears you are the end user, you should contact a reputable consultant and/or certified cisco partner.
Attempting to do by yourself will onlu cause frustration and costly mistakes.
As the other commenter commented, much of this information is on CCO. My particular company is actually using a Class-5 softswitch. The Cisco CallManager platform is what you would normally want to look further into if you are an end-user.
H.323 and SIP are just two of several protocols that allow users to use a packet network to transport streaming media such as voice and video. There are others as well such as MGCP, SCCP, IAX, H.248...
H.323 was one of the early protocols, and was derived from ISDN technology. H.323 has been around for quite some time and was adoped by many platforms such as the Avaya IP Office platform. Wikipedia has a decent write-up of it:
SIP (also documented at Wikipedia http://en.wikipedia.org/wiki/Session_Initiation_Protocol) is considered a lightweight protocol because there is more intelligence required of the endpoing where protocols such as MGCP require an endpoint to be slaved to the switch controlling endpoints. Many modern Class-5 switches are using SIP or some vendor-customized version of SIP for external communications to other switch providers.
We tend to find that no one protocol is the best and each has its benefits. If you are providing a service that is to closely resemble or emulate a PSTN POTS line then MGCP is probably a better solution. If you want to use an IP enabled phone to see line status, do conference calls, maybe some group paging then SIP may be the better answer.
Hopefully that helps out a bit!