On Fri, Jun 8, 2012 at 6:41 PM, Daniel-Constantin Mierla <mico...@gmail.com> wrote: > > On 6/8/12 1:26 PM, Aft nix wrote: >> >> On Fri, Jun 8, 2012 at 2:18 PM, Andrew Pogrebennyk >> <apogreben...@sipwise.com> wrote: >>> >>> The papers talk about transport protocol for signaling, not media/RTP. >>> I didn't hear of anyone who does RTP over TCP neither. I doubt even that >>> the performance is a primary reason behind that, for media over TCP the >>> client link must be virtually packet-loss free (due to TCP >>> retransmissions), while over UDP sometimes up to 5% packet loss can be >>> tolerated. TCP was not designed as transport for real-time media :-) >>> >>> On 06/08/2012 12:35 AM, Yang Hong wrote: >>>> >>>> Hello. >>>> >>>> SIP over TCP would reduce server performance significantly when compared >>>> with SIP Over UDP. >>>> >>>> Please read the following two papers. Combining RTP proxy with SIP over >>>> TCP would degrade SIP server performance even worse. >>>> >>>> --------------------------------------------------------------------------------------------------------------------------------------------------------------- >>>> http://www.cs.columbia.edu/~hgs/papers/Shen1008_TLS.pdf >>>> >>>> The Impact of TLS on SIP Server Performance >>>> >>>> "Securing SIP is accomplished by using TLS instead of UDP as the >>>> transport protocol. We show that using TLS can reduce performance by up >>>> to a factor of 17 compared to the typical case of SIP-over-UDP." >>>> >>>> "Network operators considering deploying SIP over TLS will need to >>>> consider the extra resources required to provide the same service >>>> quality as would be the case with UDP." >>>> >>>> --------------------------------------------------------------------------------------------------------------------------------------------------------------- >>>> >>>> http://www.cs.columbia.edu/~hgs/nossdav/2007/files/file-27-session5-paper1-nahum.pdf >>>> >>>> Evaluating SIP Proxy Server Performance >>>> >>>> "The next most signi cant performance feature is which transport >>>> protocol is used, TCP or UDP. Using TCP can reduce performance anywhere >>>> from 43 percent (the stateful proxying scenario with authentication) to >>>> 65 percent (state-less proxying without authentication). >>>> >>>> --------------------------------------------------------------------------------------------------------------------------------------------------------------- >>>> >>>> Best regards, >>>> >>>> Yang >>>>> >>>>> Date: Thu, 7 Jun 2012 13:36:39 +0200 >>>>> From: mico...@gmail.com >>>>> To: sr-users@lists.sip-router.org >>>>> Subject: Re: [SR-Users] Looking for RTP Proxy in TCP >>>>> >>>>> Hello, >>>>> >>>>> On 6/4/12 7:14 PM, Austin Einter wrote: >>>>>> >>>>>> Hi All >>>>>> Now I am using Kamailio 3.1.5 and RTP proxy 1.1. >>>>>> Looks both are compatible and working fine. >>>>>> >>>>>> The RTP Proxy basically sends/receives RTP packets over UDP. >>>>>> Is there any RTP Proxy available that does send/receive of RTP packets >>>>>> over TCP and also should be compatible with Kamailio 3.1.5. >>>>>> >>>>>> If you have any information in this regard, kindly share. >>>>> >>>>> RTP itself is specified over UDP, also I am not aware of any SIP phone >>>>> doing RTP over TCP. >>>>> >>>>> MSRP is a mechanism specified for sending message streams over TCP, we >>>>> have a module for that, but I guess is not exactly what you are >>>> >>>> looking for: >>>>> >>>>> http://kamailio.org/docs/modules/devel/modules/msrp.html >>>>> >>>>> Maybe based on it you can implement one that fits your needs. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> >> These things are wild attempt to escape some telcos blocking UDP >> packets suspecting VOIP for pushing their own IP telephony product. >> Sometimes even encrypted media is blocked because their DPI use some >> heuristic methods for detecting media packet. Like payload size in >> different codecs can give a clue about the packet. Although This bites >> in the a** of whole "net neutrality" campaign, They don't bother. >> >> These practices force VOIP providers for "obsfucation" techniques for >> escaping the DPI. Interesting thing is i know of some situation when >> they blocked VPN because people were using it to make VOIP calls. Now >> you can use VPN for other purposes! >> >> Some telco even went to lengths to block UDP "streams" by calculating >> some "threshold" bandwidth consumption. >> >> The DPI vendors are making a business case, but it all comes at price >> of making Provider inventing non standard schemes to do ordinary >> stuffs. >> >> Media over TCP is the worst idea in the history of worst ideas. But >> sometimes you have no choice. >> >> I guess big web companies should push these telcos who are afraid of >> losing their traditional TDM market share and going at lengths to stop >> media over IP. > > look at htproxy: > > http://www.mbdsys.com/foss/htproxy/file/f16c43f3c3c3/README > > it is kind of http proxy that can be used to tunnel udp packets. You need to > have a client application supporting it, on the sip server side you don't > need anything. > >
Hi Daniel, Does this tunnel over "HTTP"? I mean the actual payload goes as payload of a http packet which goes over TCP? If i'm not wrong, for sending 30 bytes of actual voice data, you have send like 1K data? Does this really work? I think i'm gonna set this up in my lab to see if it works. Thanks for the interesting link. cheers. > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - > http://asipto.com/u/katu > Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - > http://asipto.com/u/kpw > > > -- -aft _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
