Conceptual Question about Freeswitch and SIP
It's my understanding that when I call from one SIP address to another that Freeswitch manages the invites and then directly connects the two sip devices, such that freeswitch is no longer involved in the conversation. Is this how it is supposed to work? The reason for my question is that the rtp traffic is going through my freeswitch ip address at all times. It seems that both ends are speaking g711u, so I don't believe that any transcoding is going on. I must be missing something because having all data flow through Freeswitch would not be scalable and I know that one of the big differences that Freeswitch offers is scalability. What am I missing....and I know that I am:) Kurt
FreeSwitch has 3 modes of operation when it comes to sip calls. Please Note FreeSwitch is a B2BUA and not a proxy although it has proxy like modes. Mode 1) Full Media Interaction this is what you are seeing. You can completely interact with the media stream allowing for sniffing the media, transcoding, or injecting the media for various purposes. Mode 2) Media Proxy only mode. In this mode FreeSwitch just acts as a dumb RTP proxy for things like Topology Hiding and NAT busting. SIP SDPs are copied across with updates to where to send the media and a socket is set up for relaying the media. The RTP/UDPTL is just copied across the socket. Mode 3) No Media Mode. In this mode the SDPs are copied across and media is setup end to end... No media comes into freeswitch... All 3 Modes are more efficient and mode scalable then some other well known software. This is not necessarily due to the SIP stack in use, but more from a design perspective of taken from what Anthm and crew learned over the years as not to do.
From: Kurt Marasco <kmarasco@faithwork.org> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Sun, 16 Mar 2008 20:03:01 -0700 To: <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Conceptual Question about Freeswitch and SIP
It's my understanding that when I call from one SIP address to another that Freeswitch manages the invites and then directly connects the two sip devices, such that freeswitch is no longer involved in the conversation. Is this how it is supposed to work?
The reason for my question is that the rtp traffic is going through my freeswitch ip address at all times. It seems that both ends are speaking g711u, so I don't believe that any transcoding is going on.
I must be missing something because having all data flow through Freeswitch would not be scalable and I know that one of the big differences that Freeswitch offers is scalability.
What am I missing....and I know that I am:) Kurt
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Thanks for your comments Ken. They were very helpful. This group is great. I always seem to get multiple responses:) Ken Rice wrote:
FreeSwitch has 3 modes of operation when it comes to sip calls. Please Note FreeSwitch is a B2BUA and not a proxy although it has proxy like modes.
Mode 1) Full Media Interaction this is what you are seeing. You can completely interact with the media stream allowing for sniffing the media, transcoding, or injecting the media for various purposes.
Mode 2) Media Proxy only mode. In this mode FreeSwitch just acts as a dumb RTP proxy for things like Topology Hiding and NAT busting. SIP SDPs are copied across with updates to where to send the media and a socket is set up for relaying the media. The RTP/UDPTL is just copied across the socket.
Mode 3) No Media Mode. In this mode the SDPs are copied across and media is setup end to end... No media comes into freeswitch...
All 3 Modes are more efficient and mode scalable then some other well known software. This is not necessarily due to the SIP stack in use, but more from a design perspective of taken from what Anthm and crew learned over the years as not to do.
From: Kurt Marasco <kmarasco@faithwork.org> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Sun, 16 Mar 2008 20:03:01 -0700 To: <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Conceptual Question about Freeswitch and SIP
It's my understanding that when I call from one SIP address to another that Freeswitch manages the invites and then directly connects the two sip devices, such that freeswitch is no longer involved in the conversation. Is this how it is supposed to work?
The reason for my question is that the rtp traffic is going through my freeswitch ip address at all times. It seems that both ends are speaking g711u, so I don't believe that any transcoding is going on.
I must be missing something because having all data flow through Freeswitch would not be scalable and I know that one of the big differences that Freeswitch offers is scalability.
What am I missing....and I know that I am:) Kurt
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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Would FS be able to do any sort of tracking / billing in No Media Mode ? how about something like rtp timeout? On Sun, Mar 16, 2008 at 11:46 PM, Kurt Marasco <kmarasco@faithwork.org> wrote:
Thanks for your comments Ken. They were very helpful. This group is great. I always seem to get multiple responses:)
Ken Rice wrote:
FreeSwitch has 3 modes of operation when it comes to sip calls. Please Note FreeSwitch is a B2BUA and not a proxy although it has proxy like modes.
Mode 1) Full Media Interaction this is what you are seeing. You can completely interact with the media stream allowing for sniffing the media, transcoding, or injecting the media for various purposes.
Mode 2) Media Proxy only mode. In this mode FreeSwitch just acts as a dumb RTP proxy for things like Topology Hiding and NAT busting. SIP SDPs are copied across with updates to where to send the media and a socket is set up for relaying the media. The RTP/UDPTL is just copied across the socket.
Mode 3) No Media Mode. In this mode the SDPs are copied across and media is setup end to end... No media comes into freeswitch...
All 3 Modes are more efficient and mode scalable then some other well known software. This is not necessarily due to the SIP stack in use, but more from a design perspective of taken from what Anthm and crew learned over the years as not to do.
From: Kurt Marasco <kmarasco@faithwork.org> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Sun, 16 Mar 2008 20:03:01 -0700 To: <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Conceptual Question about Freeswitch and SIP
It's my understanding that when I call from one SIP address to another that Freeswitch manages the invites and then directly connects the two sip devices, such that freeswitch is no longer involved in the conversation. Is this how it is supposed to work?
The reason for my question is that the rtp traffic is going through my freeswitch ip address at all times. It seems that both ends are speaking g711u, so I don't believe that any transcoding is going on.
I must be missing something because having all data flow through Freeswitch would not be scalable and I know that one of the big differences that Freeswitch offers is scalability.
What am I missing....and I know that I am:) Kurt
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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We are able to generate cdrs in no media mode and we are able to use sip session timers to detect a dead endpoint. For obvious reasons, you can't do rtp timeout if we are not in the media path. Mike On Mar 17, 2008, at 8:51 AM, Vijay Ramnarine wrote:
Would FS be able to do any sort of tracking / billing in No Media Mode ? how about something like rtp timeout?
On Sun, Mar 16, 2008 at 11:46 PM, Kurt Marasco <kmarasco@faithwork.org> wrote: Thanks for your comments Ken. They were very helpful. This group is great. I always seem to get multiple responses:)
Ken Rice wrote:
FreeSwitch has 3 modes of operation when it comes to sip calls. Please Note FreeSwitch is a B2BUA and not a proxy although it has proxy like modes.
Mode 1) Full Media Interaction this is what you are seeing. You can completely interact with the media stream allowing for sniffing the media, transcoding, or injecting the media for various purposes.
Mode 2) Media Proxy only mode. In this mode FreeSwitch just acts as a dumb RTP proxy for things like Topology Hiding and NAT busting. SIP SDPs are copied across with updates to where to send the media and a socket is set up for relaying the media. The RTP/UDPTL is just copied across the socket.
Mode 3) No Media Mode. In this mode the SDPs are copied across and media is setup end to end... No media comes into freeswitch...
All 3 Modes are more efficient and mode scalable then some other well known software. This is not necessarily due to the SIP stack in use, but more from a design perspective of taken from what Anthm and crew learned over the years as not to do.
From: Kurt Marasco <kmarasco@faithwork.org> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Sun, 16 Mar 2008 20:03:01 -0700 To: <freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Conceptual Question about Freeswitch and SIP
It's my understanding that when I call from one SIP address to another that Freeswitch manages the invites and then directly connects the two sip devices, such that freeswitch is no longer involved in the conversation. Is this how it is supposed to work?
The reason for my question is that the rtp traffic is going through my freeswitch ip address at all times. It seems that both ends are speaking g711u, so I don't believe that any transcoding is going on.
I must be missing something because having all data flow through Freeswitch would not be scalable and I know that one of the big differences that Freeswitch offers is scalability.
What am I missing....and I know that I am:) Kurt
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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Hi Vijay,
Would FS be able to do any sort of tracking / billing in No Media Mode ? how about something like rtp timeout? Yes and no, respectively, although the same effect as the latter can be achieved using a SIP session timer - have a look at conf/sip_profiles/default.xml
--Dave
On Sun, Mar 16, 2008 at 11:46 PM, Kurt Marasco <kmarasco@faithwork.org <mailto:kmarasco@faithwork.org>> wrote:
Thanks for your comments Ken. They were very helpful. This group is great. I always seem to get multiple responses:)
Ken Rice wrote: > FreeSwitch has 3 modes of operation when it comes to sip calls. Please Note > FreeSwitch is a B2BUA and not a proxy although it has proxy like modes. > > Mode 1) Full Media Interaction this is what you are seeing. You can > completely interact with the media stream allowing for sniffing the media, > transcoding, or injecting the media for various purposes. > > Mode 2) Media Proxy only mode. In this mode FreeSwitch just acts as a dumb > RTP proxy for things like Topology Hiding and NAT busting. SIP SDPs are > copied across with updates to where to send the media and a socket is set up > for relaying the media. The RTP/UDPTL is just copied across the socket. > > Mode 3) No Media Mode. In this mode the SDPs are copied across and media is > setup end to end... No media comes into freeswitch... > > All 3 Modes are more efficient and mode scalable then some other well known > software. This is not necessarily due to the SIP stack in use, but more from > a design perspective of taken from what Anthm and crew learned over the > years as not to do. > > > > >> From: Kurt Marasco <kmarasco@faithwork.org <mailto:kmarasco@faithwork.org>> >> Reply-To: <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> >> Date: Sun, 16 Mar 2008 20:03:01 -0700 >> To: <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> >> Subject: [Freeswitch-users] Conceptual Question about Freeswitch and SIP >> >> It's my understanding that when I call from one SIP address to another >> that Freeswitch manages the invites and then directly connects the two >> sip devices, such that freeswitch is no longer involved in the >> conversation. Is this how it is supposed to work? >> >> The reason for my question is that the rtp traffic is going through my >> freeswitch ip address at all times. It seems that both ends are speaking >> g711u, so I don't believe that any transcoding is going on. >> >> I must be missing something because having all data flow through >> Freeswitch would not be scalable and I know that one of the big >> differences that Freeswitch offers is scalability. >> >> What am I missing....and I know that I am:) >> Kurt >> >> _______________________________________________ >> Freeswitch-users mailing list >> Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >
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On Mar 16, 2008, at 10:03 PM, Kurt Marasco wrote:
It's my understanding that when I call from one SIP address to another that Freeswitch manages the invites and then directly connects the two sip devices, such that freeswitch is no longer involved in the conversation. Is this how it is supposed to work?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall
The reason for my question is that the rtp traffic is going through my freeswitch ip address at all times. It seems that both ends are speaking g711u, so I don't believe that any transcoding is going on.
Well in some cases you can do p2p but since no transcoding is going on the only limit is the amount of context switches your machine can take on. I know you can do way more than you think doing media thru FreeSWITCH.
I must be missing something because having all data flow through Freeswitch would not be scalable and I know that one of the big differences that Freeswitch offers is scalability.
Actually it can scale with media just not as far. If you set the bypass_media=true variable before you bridge we'll get out of the media path. There are instances where you want to always be in the media path. This paired with late negotiation can give you access to the SDP and even allow you to choose the mode on the fly. /b
What am I missing....and I know that I am:) Kurt
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Thanks for your comments, that was a tremendous help. So is it typically bandwidth or computing power that limits scalability? And I know that is a somewhat loaded question. I'm initially looking at setting up a relatively small environment for myself friends and family to utilize and in the process save some money on phone bills and use it as a lab for me to learn more about voip. I'm considering keeping FS behind my home broadband or putting it on a fairly limited power VPS. At home I would have much computing power, but relatively little bandwidth. My computing power would be limited on a VPS , but much higher bandwidth. Kurt Brian West wrote:
On Mar 16, 2008, at 10:03 PM, Kurt Marasco wrote:
Well in some cases you can do p2p but since no transcoding is going on the only limit is the amount of context switches your machine can take on. I know you can do way more than you think doing media thru FreeSWITCH.
participants (6)
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Brian West
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David Knell
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Ken Rice
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Kurt Marasco
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Michael Jerris
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Vijay Ramnarine