4 Jun
2012
4 Jun
'12
1:14 p.m.
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
7 Jun
7 Jun
7:36 a.m.
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
6:35 p.m.
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-…
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(a)gmail.com
To: sr-users(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 
8 Jun
8 Jun
4:18 a.m.
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-…
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(a)gmail.com
To: sr-users(a)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:
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
> 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 
7:26 a.m.
On Fri, Jun 8, 2012 at 2:18 PM, Andrew Pogrebennyk
<apogrebennyk(a)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:
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
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-…
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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users Date: Thu, 7 Jun 2012 13:36:39 +0200
From: miconda(a)gmail.com
To: sr-users(a)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:
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
> 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
8:41 a.m.
On 6/8/12 1:26 PM, Aft nix wrote:
On Fri, Jun 8, 2012 at 2:18 PM, Andrew Pogrebennyk
<apogrebennyk(a)sipwise.com> wrote:
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
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:
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. 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-…
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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users Date: Thu, 7 Jun 2012 13:36:39 +0200
From: miconda(a)gmail.com
To: sr-users(a)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
>
9:23 a.m.
On Fri, Jun 8, 2012 at 6:41 PM, Daniel-Constantin Mierla
<miconda(a)gmail.com> wrote:
On 6/8/12 1:26 PM, Aft nix wrote:
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.
On Fri, Jun 8, 2012 at 2:18 PM, Andrew Pogrebennyk
<apogrebennyk(a)sipwise.com> wrote:
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.
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:
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.
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-…
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(a)gmail.com
> To: sr-users(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users 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
10:27 a.m.
Hi,
On 06/08/2012 03:23 PM, Aft nix wrote:
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
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?
4585
days inactive
4589
days old
7 comments
6 participants
participants (6)
-
Aft nix -
Andreas Granig -
Andrew Pogrebennyk -
Austin Einter -
Daniel-Constantin Mierla -
Yang Hong