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. Thanks a lot. Austin
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 -- 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
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-n... 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: miconda@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
-- 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
_______________________________________________ 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
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-n...
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: miconda@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
On Fri, Jun 8, 2012 at 2:18 PM, Andrew Pogrebennyk <apogrebennyk@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-n...
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: miconda@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. Cheers. -- -aft
On 6/8/12 1:26 PM, Aft nix wrote:
On Fri, Jun 8, 2012 at 2:18 PM, Andrew Pogrebennyk <apogrebennyk@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-n...
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: miconda@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. 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
On Fri, Jun 8, 2012 at 6:41 PM, Daniel-Constantin Mierla <miconda@gmail.com> wrote:
On 6/8/12 1:26 PM, Aft nix wrote:
On Fri, Jun 8, 2012 at 2:18 PM, Andrew Pogrebennyk <apogrebennyk@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-n...
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: miconda@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
Hi, On 06/08/2012 03:23 PM, Aft nix wrote:
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.
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?
Another approach we're using for example to bypass restrictive hotel Wifis is having a VPN connected via TCP port 443, where you can tunnel any kind of data then. I haven't seen anyone blocking this kind of traffic, but who knows... Andreas
participants (6)
-
Aft nix -
Andreas Granig -
Andrew Pogrebennyk -
Austin Einter -
Daniel-Constantin Mierla -
Yang Hong