Is Asterisk involved on communications between 2 SIP Phones?

Hi,

I know that OpenSER is a stateless SIP Proxy, that means that if I have a conversation between 2 SIP Phones on the same local network, THERE is no traffic on the SIP Proxy server, and all the traffic takes part between the phones… So there is no load on the SIP Proxy Server.

NOW… The Asterisk part… If I am using Asterisk as a SIP Server, I heard it acts as a statefull SIP Proxy, taking responsability on the communicaitons. Does that means that it acts as “Midlle in the man” server, and all the traffic, including RTP traffic, has to go trhought it ?

That it means that a conversation between 2 phones on the same network is managed and gestionated by Asterisk, and therefore, the server has to asume that LOAD ?

Thks for your clarifications

Yes. Asterisk stays in the middle of all calls. All RTP flows through Asterisk. There are disadvantages (e.g. load on the server), but there are advantages (e.g. ability to record the RTP streams, Asterisk can process DTMF signalling during the call, three-way calling, etc.).

You have to budget for the CPU on the Asterisk server to handle the streams flowing through it.

want to just add that you don’t have to let Asterisk handle the media stream. if you use a reinvite, then Asterisk steps out, but as mentioned, you lose the advantages … but save CPU.

Thks for your reply, now I understand it…

Just a clarification… How do I use a reinvite on Asterisk to avoid that flod ?

And also… On OpenSer (SIP Stateless Proxy), there is no such as flood, right ?

in sip.conf, set “canreinvite=yes” for the user. Asterisk won’t issue a reinvite if you use Dial options that require Asterisk to stay in the stream though.

in that way, it acts more like OpenSER. how many calls are you talking about ?

Thsk a lot…

BTW… What do you mean by “Dial Options” that require the call to stay on the flood ? Also… on that setup…Would be the user be able to “capture” the call from another phone/extension when is ringing on other phone, presing a keyboard combination ?

I am asking so because I am talking about an enviroment of 700 users.
My customer wants to have ONLY one Asterisk Server, and only 1 E1 line, since they would do a very few external calls (not more than 20 simultaneously). They would use LinkSys phones and I am not sure how much internal traffic it could be, but 700 users seems a lot for a single Asterisk machine, not ?

That´s why I was thinking about either putting OpenSER in front, or either use that reinvite option you mention…

haven’t you asked this question already on the -users list ?

700 registrations shouldn’t be a problem for a (min) dual-CPU machine properly configured with a decent amount of resources. the big question is the number of simultaneous calls, and whether you’re doing things like transcoding and recording.

i’m guessing you can get some idea of the number of calls they handle ?

if using reinvites, Asterisk won’t step out of the path until the call is setup and answered … so pickup shouldn’t be a problem. you’ll lose the option of Asterisk recording aspects of transfers etc.

the Dial options are things like TtWw where Asterisk needs to be in the media path to hear them.

[quote=“baconbuttie”]
if using reinvites, Asterisk won’t step out of the path until the call is setup and answered … so pickup shouldn’t be a problem. you’ll lose the option of Asterisk recording aspects of transfers etc.[/quote]

Other things that you lose by releasing the media from Asterisk with reinvites:

No call detail records.
No hinting.
No sla (if Asterisk ever gets that working in 1.4.x).

Essentially, with reinvites, Asterisk just acts as a registrar, not a PBX. SIP UA’s (i.e. phones) can make peer-to-peer calls which Asterisk is completely unaware of.

[quote=“diiguy”]

Other things that you lose by releasing the media from Asterisk with reinvites:

No call detail records.
No hinting.
No sla (if Asterisk ever gets that working in 1.4.x).

Essentially, with reinvites, Asterisk just acts as a registrar, not a PBX. SIP UA’s (i.e. phones) can make peer-to-peer calls which Asterisk is completely unaware of.[/quote]
You still get CDRs and hints even if Asterisk drops itself out of the media path.

[quote=“srl100”]
You still get CDRs and hints even if Asterisk drops itself out of the media path.[/quote]
How would you enforce keeping Asterisk in the signalling path while releasing only the media?

[quote=“baconbuttie”]haven’t you asked this question already on the -users list ?
[/quote]

Yes, I did, but it was not exactly the same question. The question on the users list was why some post on digium talks about 240 simultaneos calls on ABE, while others mention not more of 80, and some details about how Asterisk can scale.

Question here was about using a SIP PROXY behavior to redude load…
I mentionated estimated load only when I was asked…

I knowns that questions were very similiar, but reasons were different…

BTW, thks for all this replys, I am learning a lot… this is very interesting.

[quote=“diiguy”][quote=“srl100”]
You still get CDRs and hints even if Asterisk drops itself out of the media path.[/quote]
How would you enforce keeping Asterisk in the signalling path while releasing only the media?[/quote]
I’ve got ‘canreinvite=yes’ on both caller and callee - as soon as the call is setup and Asterisk sees that the endpoints have codecs in common it sends a reinvite to each party with the media information (SDP) of the other, so it drops itself out of the media path, but the call control still gets routed via the Asterisk server.