VoIP and VoIP Topology Questions


I have a couple basic questions in regards to VoIP protocols and VoIP network topology, and please excuse me if they are basic. I have setup an Asterisk 1.6.2 build with libpri, libss7, openr2, the latest DAHDI drivers for a Digium TE420 4 port E1 card, and I’m using Ubuntu Linux 9.1. I have SS7 connections from the PSTN connected to the TE420. I also have a router setup that connects between the PC and 4 SIP phones. Everything is configured correctly and working great. However, I have a few questions.

  1. In my configuration, what would be operating as the VoIP Gateway? Through my research on VoIP so far, it looks like the VoIP Gateway acts as the PSTN Gateway. Is the TE420 PCI card handling the translation from TDM circuit switched data to VoIP (SIP in this case) packet data?

  2. In my configuration, would the PC that Asterisk is built on be operating as the proxy server? I’ve seen both proxy server terminology used as well IP PBX Server.

  3. I understand from my research that voice data (being streaming data in nature) needs to be handled differently than traditional packet data over IP. As a result, various VoIP protocols were developed (e.g. SIP, H.323, IAX, SCTP, etc.) to allow voice data to be transferred over IP; hence, the name VoIP. Latency with traditional packet data can be accepted because the data still arrives in tact. Latency with voice data (streaming data) cannot be tolerated (150ms is acceptable but much more than 300ms and you start having issues) because then the conversation degrades. My question is this: where does the transition from a particular VoIP protocol to traditional IP take place within the VoIP network? I ask this because I want to phase out an old hardware PBX and move into an Asterisk VoIP PBX. Before I completely phase out the old PBX, I’d like to connect it to Asterisk but can I use traditional IP or do I have to use a VoIP protocol such as SIP? I’m thinking the latter due to my explanation above about handling the voice data. I guess I’m having a mental block about VoIP protocols being needed to handle voice data but this data eventually goes over a traditional IP network over the internet until it reaches the far end. Where do the two meet and get exchanged and/or translated?

Thanks for your help in advance. Please direct me to a link or book if you know of one that will help me understand the questions I have or feel free to share your knowledge/experiences. I appreciate it!

  1. the whole system
  2. no
  3. SIP is not a VoIP protocol from the point of view of crtiical peformance, only RTP is is for SIP connections.