Having tinkered with SIP and H.323 for a while, I'm wondering about the true technological advance represented by Skype - or is it just cleverly packaged hype to induce users to adopt the product, and later charge them for PSTN connectivity?
The distributed directory of Skype is probably more efficient than SIP proxies that could be located dozens of hops away from my home PC (for example on the East Coast in the case of FWD). Call establishment should be faster as a result (note: I'm not a Skype user yet, I'm only making guesses).
Now looking at the PSTN, SS7 provides an effective solution for progressive address (telephone number) resolution, each central exchange contributing to the routing of the call signalling. Where is the innovation in Skype?
The claim about Skype implementing a better, proprietary, audio codec is dubious at best. G.723 and G.729, as well as other CELP based codecs, provide good compressed speech. An issue is that most users have only experienced the free GSM 6.10 codec in open source VoIP implementations. Skype may well sound better than GSM, but does it beat G.723 and G.729? - which have the added benefit of being standards.
But my main problem is in understanding what is the true meaning of the P2P nature of Skype. After all, once a SIP call is established, RTP media communication occurs directly between the two endpoints (e.g. soft-phones) of the call. What is the advantage of Skype over SIP/RTP?
Skype might have more focus on providing easy NAT/firewall traversal compared to SIP. However, SIP offers several draft recommendations on this same subject, most notably STUN. Setting up my XTen soft-phone for connection to FWD behind our company NAT/firewall wasn't easy, but neither was it impossible.
Pending more technical information about Skype, it seems to me that the main technology breakthrough has been made in making Skype easy to install and NAT/firewall friendly while re-using existing characteristics of proven, open, VoIP solutions.
Skype might be a really good product but I doubt it will survive in the heavily standards-oriented telephony world. But that's only my guess:)
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The distributed directory of Skype is probably more efficient than SIP proxies that could be located dozens of hops away from my home PC (for example on the East Coast in the case of FWD). Call establishment should be faster as a result (note: I'm not a Skype user yet, I'm only making guesses).
Now looking at the PSTN, SS7 provides an effective solution for progressive address (telephone number) resolution, each central exchange contributing to the routing of the call signalling. Where is the innovation in Skype?
The claim about Skype implementing a better, proprietary, audio codec is dubious at best. G.723 and G.729, as well as other CELP based codecs, provide good compressed speech. An issue is that most users have only experienced the free GSM 6.10 codec in open source VoIP implementations. Skype may well sound better than GSM, but does it beat G.723 and G.729? - which have the added benefit of being standards.
But my main problem is in understanding what is the true meaning of the P2P nature of Skype. After all, once a SIP call is established, RTP media communication occurs directly between the two endpoints (e.g. soft-phones) of the call. What is the advantage of Skype over SIP/RTP?
Skype might have more focus on providing easy NAT/firewall traversal compared to SIP. However, SIP offers several draft recommendations on this same subject, most notably STUN. Setting up my XTen soft-phone for connection to FWD behind our company NAT/firewall wasn't easy, but neither was it impossible.
Pending more technical information about Skype, it seems to me that the main technology breakthrough has been made in making Skype easy to install and NAT/firewall friendly while re-using existing characteristics of proven, open, VoIP solutions.
Skype might be a really good product but I doubt it will survive in the heavily standards-oriented telephony world. But that's only my guess :)
Serge Kruppa, Simitel