现在大家搞VOIP是H323还是SIP的多？

zhangjiakouzf

sip主要是协议简单。容易扩展，但缺乏严格标准

servasyy

sip的扩展性极强，标准也是逐渐完善的，前景上，当然是sip

chenhawk

现在H.323是主流，但用不了多久SIP就会成主流。现在大的VoIP厂商的产品都开始支持SIP了

aloading

VoIP技术的发展有两大方向，一是遵循国际电联H.323标准，二是遵循IETF的SIP协议。国际电联的H.323基本上是一种兼顾传统呼叫流程和IP网特点发展而来的成熟的开放标准体制，代表了VoIP的大潮流。它的特别之处是吸取了许多电信网的组网、互联和运营经验，能够与PSTN网，以及其他数据业务和应用网互联互通。这正是自1995年以来，H.323标准在全世界广泛使用的一个重要原因。采用H.323技术体制,VoIP运营商可以基本上继承传统运营商的管理和运维模式--这对中国以及东南亚国家组建VoIP大网特别重要。在中国，运营商组建的VoIP网都是全国性大网，而且对网络的扩展性和稳定性要求极高，组网必须多层多域，覆盖城市多达400个，每月的话务量在几亿分钟以上。组建这样的大网，采用H.323技术是合适的选择。一些新技术可能在数据与语音的结合上做得比较好,但是它们对于如何组网、运营、管理却考虑得比较少。  

　　VoIP技术的另一个发展趋势是遵循IETF倡导的SIP协议。SIP协议在美国十分热门，近来发展很快。这是因为：①美国的互联网技术发展很快; ②SIP技术发展初期的实现条件比较简单。在美国充分开放的电信环境下，"简单"能带来巨大的创新诱惑和想像空间，并由此吸引了美国国内一批设备制造商和新兴运营商、服务商投入大量财力和精力。SIP技术基本上是基于动态Internet模式来建网的，它的网络不是一个多层次的网络，而是一个扁平的单层次网络，呼叫流程与PSTN网有所不同。例如，用户呼叫上海地区号码021，那么在以H.323标准组建的VoIP网上，它首先考虑如何将这个呼叫送到上海，然后再送达目的地。其寻址、组网方式与传统电话网极其相似。但在SIP网上则靠动态数据库的方式来寻址，甚至可以没有长途和短途之分。SIP模式的优点是与Internet紧密结合，适于开发新的、与互联网结合的语音应用；其缺点是在组网、管理、运营、计费方面的考虑还有待成熟，在与传统PSTN网的互联互通方面对一些非正常情况的处理还有待完善，在组建VoIP大网的实践方面，还有待积累经验。  

　　应该说，在中国和东南亚国家选用H.323技术标准来组建VoIP网是适宜的。这是因为，这些国家的电话普及率并不高，基本通话业务的需求还在快速增长。这就意味着在相当长的一段时间内，话音业务仍会按传统的运营和管理体制来运行，这就需要VoIP网具有与传统电信网接近的呼叫流程和处理机制，具有很好的扩容性、可管理性、可运营性，与现有的PSTN网有着较好的互通性。美国国情与中国不同，美国的电话普及率已达90%，互联网发展十分迅速，因此在VoIP上的策略是更注重作为互联网的增值应用。要知道，组建一个互联网增值业务网和一个电信级VoIP大网的概念是有所不同的。目前，最大的基本业务VoIP网络在中国。

mars531706

Our company's sorfswitch platform is based on SIP.

hoowa

SIP是新协议。还在发展中。

flyingnn

我在用SIP，VOCAL SERVER  ，LINKSYS 的终端，公司用。

pwestly

SIP
现在已经没有人怀疑其前景了。应该也不会有公司只做H323了。

matthew6868

来自http://www.gnugk.org/sip.html上的一段。。。。

Will SIP take over and will H.323 die out ?
Probably not. While SIP is very strong in the consumer market, H.323 has a very strong position in the telco market. Both for good reasons.
Here is a quote from Simon Horn on the GnuGk mailinglist in February 2006:

[...] Why haven't the major telcos switched to SIP? 2 reasons
1. LRQ  There is no equivalent in SIP. You cannot deploy a large SIP server
farm network anywhere as efficiently as huge H.323 GK hierarchy
2. ISDN (Traditional Phone network). ISDN is H.320 and is designed for and
is far more compatible with H.323

However I will point out that H.323 does have better NAT traversal
capabilities (actually it does) and GNUGK NAT traversal method is far
better than anything capable in SIP and even within the ITU standard
H.460.18/19. There is also the ability to just simply port forward the
required ports in the router, this is more problematic is SIP..  The
reason, basically the design of the protocol, In SIP the receiving and
sending ports for RTP must be the same which means that port translation in
symmetric NATs (more than 50% of the market) is a major problem so STUN
becomes basically useless, you have to revert to ICE which requires up to
10 extra messages to traverse the NAT which does take some time to
negotiate and for business needs it becomes unusable. This is why your ITSP
insists you buy an expensive box to connect your phone to .  The GNUGK
method uses 0 extra messages and instant.. Though I will point out that
some SIP servers such as SER do support a method similar to GNUGK but not
as efficient

Another major problem is securing SIP. I have spent quite a bit of time
looking at porting our security ideas to SIP and have come to one
conclusion, it can't be done without upgrading an entire Network to support
it. The basic problem is.
1. Lack of a dedicated End-to-End Calling channel (H.225.0 Signalling)
which makes it Impossible for 2 parties in a call to authenticate each
other directly. You need to rely on the servers and things like SIP
Identity (which requires 6 extra messages and can take up to 2 sec just to
identify the callers). In H.323 you just add a cryptoToken in the H.225
Setup message and your done. No extra messages, almost no extra delays)
2. Lack of a equivalent H.225 Call Proceeding Message which means its
impossible to do the encryption setup before the remote party rings. Which
means that you have up to 3 sec of 'Dead Air" when you pick the phone up.
For business this is unacceptable. They also cannot agree on which SRTP
method to use MiKey or sdesc so a lot of these secure phones/networks can
not interperate. In H.323 you can setup an encrypted call in under 400 ms
and it will be 100% interoperable with almost every piece of legacy H.323
equipment. In SIP you have to upgrade everything. If you pick the wrong
method MiKey or sdesc you may have to do it twice.

My point is from a programmers/implementation point of view H.323 is still
the only complete solution. Yes SIP is very good in it's box but outside it
it's not as good and for large scale business needs its not quite up to it.

[...] SIP is 10 years old, 5 years ago they were saying H.323 is finished SIP
will rule, well its 5 years later and H.323 is still here and the people
who were saying that are moving onto Asterix. My bet is that in another 5
years H.323 will still be around and the Asterix people will be moving onto
something else.  Open Source SKYPE clone perhaps

Some light reading on SIP
http://corporaterat.blogspot.com ... ective-painful.html
http://www.dailypayload.com/2006/0130.html

Simon

上面这两篇文章也值得一读：
Celebrating SIP's 10th Birthday
http://www.dailypayload.com/2006/0130.html

Developer Perspective: The painful evolution of SIP
http://corporaterat.blogspot.com ... ective-painful.html

我在想H.323和SIP同样是十岁（从1996年开始），为什么差距会这样？SIP真的会是未来吗？难道不会有新的协议替代SIP？
ITU已经着手开始制定第三代多媒体系统H.325[http://www.packetizer.com/ 这里有一篇它的介绍]，而SIP呢？也许H.323会消失，但是替代它的我认为不会是SIP。虽然我是做H.323的，不是很了解SIP，以前一直都认为SIP比H.323简单。当我看了上面两篇文章，我已经不这么认为了。当然这些也只是我所了解和我所认为的，仅仅代表我的观点。

[ 本帖最后由 matthew6868 于 2007-1-17 13:10 编辑 ]

sunkits

我们公司的VoIP终端同时支持SIP 和H323,客户还是用H323的多,
不过SIP发展得很快.