Planning a large installation with SER or OpenSER

Improving performance

• Store the voice prompts in RAM instead of harddisk: Take a look at ramfs. All you need then is to create a link (ln -s) in /var/lib/asterisk/sound to then ramdrive you created using ramfs.

SIP proxy before Asterisk (e.g. SER)

SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic.

Howto: Getting to know SER

If you want a good explaination of SER and how to use it start here:
Latest Guide http://www.iptel.org/ser/doc/gettingstarted
Old Guide http://siprouter.onsip.org/doc/gettingstarted/
They have GREAT pre-written configs and walk you through every part of SER. I was about to scrap SER before I found these tutorials.

And then there's the OpenSER documentation on how to integrate with Asterisk in realtime for VoiceMail and MWI. Find even more on the same topic here.

Benefits for Asterisk

We are using both, SER and Asterisk, in heavy production environment. SER is the BEST SIP proxy that i found. But it is just sip proxy. It can serve a _LOT_ connections (10,000 users, 20 cals per second). Asterisk is more like telephone switch with lot of features, but far slower.

One nice feature of SER is that users can set up their own SIP accounts using a web interface and not needing to edit *.conf files.

Q: In a large installation (many users) - how do you off-load Asterisk by managing RTP traffic effectively?
A: There's no single truth here, but here's my opinion:

• If you are all on the same internal network, make sure the SIP phones support re-invites and use that.
• If you have users all over the Internet, use a SIP proxy (like SER) as a front-end to Asterisk. You will still be forced to handle a lot of RTP streams (because of NAT), but can distribute that over a SIP-proxy network with SRV records, DNS round-robin techniques or forcing the users to register with different proxies.

Example: FWD and SIPGATE both run SER coupled with Asterisk

There's been a couple of suggestions that we should make Asterisk a good SIP proxy. If you spend some time learning to understand Asterisk's architecture, you'll also understand that this would not really work. I'm not saying the SIP channel can't be improved, I'm just saying that it has to work with the rest of Asterisk's architecture. Asterisk in combination with a separate SIP proxy is a very powerful solution.

With a SIP proxy we have the following scenario:

• Alice's UA sends an INVITE to bob@domain
• The proxy responsible for this domain receives this and looks up bob in a user location or alias table
• The proxy FORWARDS the same INVITE to bob@currentlocation, maybe several different locations
• When Bob answers somewhere, the proxy cancels the call to the non-answering locations and forwards the OK to Alice
• Alice ACKs the OK to bob and the call is UP

In this scenario, there's only one SIP dialogue, between Alice and Bob with the proxy in the middle of signalling, but acting as a proxy and not as a user agent (the proxy can't and should not answer or originate calls).

A quote

Asterisk and SER Diagram

Here is a nice broken link(?) of how to use SER and Asterisk together.

MWI - message waiting indication

See also Asterisk MWI for related info

Method 1

I don't use Asterisk as a UA, no SIP users are registered to Asterisk, instead all that is handled by SER. So Asterisk just needs to be able to reach the phone. For this you need a "peer" statement for each phone in sip.conf (which could be created based upon the SER registry data and using ast_data / res_data). Note that this peer never takes calls and is solely used for MWI purposes.

So we just need a way for Asterisk to contact the phone when it has a MWI notification (host=phone_ip). Yet another way to supply this IP is through with DDNS records.

In any case you will need a peer like this - for each phone:

 [2114]
 type=peer
 username=2114
 insecure=yes                   
 canreinvite=no                 
 context=default
 mailbox=2114
 host=SIP003094C274B3.bna01.isdn.net

Method 2 - by Andreas Granig

Q: I assume that asterisk would have to send the SIP NOTIFY message to the registered SIP clients via ser. So will I have to create a custom Asterisk and ser modules to get this to work?
A: Asterisk sends NOTIFY only to UACs that are registered at the Asterisk. If you just forward voicemail calls to Asterisk and the UACs are registered at SER, you have to use the 'externnotify'-option in voicemail.conf of Asterisk. We use a script wrapped around sipsak to turn on/off MWI by sending proper NOTIFY messages.

Details by Paul (aka Java Rockx): I use sipsak to send the message from my Asterisk server to my ser proxy. I use the Asterisk "externnotify" in the voicemail.conf file to specify an external bash script. This script then simply "touches" a file in /var/spool/mwi/ for later processing by cron.

My cron job comes along after the fact and picks up all the files in /var/spool/mwi/ and generates the appropriate SIP NOTIFY message and then it uses sipsak to "shoot" it to the ser proxy. Note that asterisk is a registered UA with ser. Doing this I assumed I could simply use sipsak from the asterisk server to send NOTIFY messages to the ser box.

This works really well for me when I need to turn on the MWI for a UA. What I still can't do is turn the MWI off when there are no new messages. I'm really having a hard time trying to determine how to hook the "Hang-Up" event in Asterisk. I need to essentially have an "externnotify" for hang ups so I can check the message status for a the mailbox and cancel the MWI if there are no new messages.

I also need to handle SUBSCRIBE messages that the SIP proxy gets. I'm planning on have the ser.cfg file execute an external script that will get a message to the cron job on my Asterisk box. Once the Asterisk box gets notified of the SUBSCRIBE message, it will process the same as if the user just had a voice mail left and the externnotify event got fired.

...and here is the solution!

Method 3

Q: If you have your SIP phones registered with SER but your voicemail is handled by asterisk, how do you get the MWI (Message Waiting Indicator) light to function on the phone?

A: In sip.conf create a section pointing at your SER router.

 [ser]
 type=friend                ; We allow incoming and outgoing calls. Use peer if you are only doing MWI
 context=ser                ; This is the context incoming calls land in
 host=ser.server.tld        ; This is the hostname or IP address of your SER server
 fromdomain=ser.server.rld  ; This is your SER_DOMAIN
 insecure=very              ; This allows incoming calls from the phones routing through ser to be passed into asterisk
 mailbox=user@context       ; This is where you list the voicemail boxes to monitor

This tells asterisk that if a voicemail comes in to "user" then it needs to send a SIP NOTIFY message to the "ser.server.tld" phone. Well this is all well and good except how does SER deliver this NOTIFY to the phones? First thing is that you need to make a tiny change to the asterisk code to pass the mailbox user in the SIP NOTIFY packet.

--- channels/chan_sip.c.orig    Thu Jul 14 12:03:18 2005
+++ channels/chan_sip.c Thu Jul 14 12:05:26 2005
@@ -9710,6 +9710,7 @@
        /* Called with peerl lock, but releases it */
        struct sip_pvt *p;
        int newmsgs, oldmsgs;
+       char *s;

        /* Check for messages */
        ast_app_messagecount(peer->mailbox, &newmsgs, &oldmsgs);
@@ -9735,6 +9736,10 @@
        /* Recalculate our side, and recalculate Call ID */
        if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip))
                memcpy(&p->ourip, &__ourip, sizeof(p->ourip));
+       strcpy(p -> username, peer -> mailbox);  /* Username = Mailbox name */
+       s = strchr(p -> username, '@');          /* Remove the context part */
+       if (s != NULL)
+                *s = 0;
        build_via(p, p->via, sizeof(p->via));
        build_callid(p->callid, sizeof(p->callid), p->ourip, p->fromdomain);
        /* Send MWI */

After this patch is applied, the MWI NOTIFY messages coming from asterisk will have the URI user@ser.server.tld. This can be then routed with ser to the correct phone with normal SER routing rules. ie. SER does a lookup("location") and then a t_relay(). I don't believe this patch should effect any non-ser controlled sip phones.

For me, this method was a lot easier then Method 2 listed above. You can add as may mailbox's as you like into the mailbox= line in the asterisk sip.conf file. One possible problem is if you have a mailbox called 1000@cx1 and another called 1000@cx2, this patch will make the MWI indicator light up for phone 1000@ser.server.tld when either mailbox gets a message. A simple modification to the patch and SER could be used to handle multiple contexts if required however this simplification is sufficient for me.

Method 4 - using realtime (ARA) and loading all SIP peers at Asterisk startup

Q: We use the Asterisk RealtimeArchitecture for voicemail users and SIP peers. The database table for the sip peers is a view from the OpenSER subscriber table.
The MWI for a user will only work, if the user object (sip peer) is loaded into memory and visible with the CLI command "sip show peers". This happens, for example, if the user call his own mailbox for checking voicemail.
Is their a possibility to load all peers from database on asterisk startup so that the users will get their NOTIFY without calling their own mailbox first?

A: See here: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/134846

A: If the phone wants MWI notifications, it should send a SUBSCRIBE with header "Event: message-summary". That should be routed to Asterisk, which will make ut load the peer in memory. Starting with the config from Realtime Integration Of Asterisk With OpenSER, you can add:

    if(is_method("SUBSCRIBE")
            && search("^Event: message-summary"))
    {
        # if there is no R-URI username, grab From URI
        if(!uri=~"sip:.+@")
        { # add From username as R-URI username
            avp_pushto("$ruri/username","$fU");
        }
        # fix some broken subscriptions
        if(!search("^Accept: application/simple-message-summary"))
            append_hf("Accept: application/simple-message-summary\r\n");
        setdsturi("sip:__asterisk_ip__:__asterisk_port__");
        t_relay();
        exit;
    }

Method 5

Assuming you have a simliar setup to method 3 where Asterisk handles voicemail, and you don't need to reach the phone from Asterisk.. It is possible to have MWI sip notify messages sent to SER without patching. In Asterisk's sip.conf, use host field for each phone to send sip notify messages to SER.
Example:
SER server - sip.zxy.com
In sip.conf:
[phonea]
..
host=sip.zxy.com
..

The setup of your dial plan may matter for this to work. In some cases it wasn't necessary to change the dial plan. If you find calls getting routed back to SER (eg 2 missed calls instead of 1), then change your Dial plan/macro for the phone.
Example:
exten => 2222,1,Voicemail(2222@default)

Example installation

Working with both Asterisk and SER:
1. Configure diferent sip listening ports for SER (/etc/ser/ser.cfg) and Asterisk (/etc/asterisk/sip.conf).
2. Configure SER (/etc/ser/ser.cfg) for forwarding calls based in destination. For example adding:

       if (uri=~"^sip:0[0-9]*@yourdomain") {
               forward( 10.10.10.10, 5070 );  //Where local asterisk is listening
               break;
       }

Asterisk and SER on same box

(Documentation in SER admin guide: http://www.iptel.org/ser/doc/seruser/seruser.html)
3. Configure Asterisk as PSTN gateway. I haven't experience in this point.

Intro on differences between Asterisk and SER

by Alex Barnes

Load balancing with SER

See discussion here: http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/162042
Partial quote:

 "SER/OpenSER can get around call forwarding/transfer problems.  You just need
 to account for those SIP dialogue's that can be problematic, and bypass
 using the dispatcher module for those situations.  

 One thing to remember is to replicate usr-loc info that is cached in memory,
 otherwise load-balancing for INVITE's will fail if passed to a proxy that
 did not receive the REGISTER request for sending party.  The standard
 t_replicate() function only supports replication between two SER/OpenSER
 proxies, so if you are looking to add in any more, you will need a more
 manual approach.  OpenSER 1.1.0 actually offers the ability to operate
 completely out of a database, even for usr-loc, which could work for you.

 Other approaches for load balancing include DNS round-robin techniques, and
 separate software or hardware appliances dedicated to load distribution.  I
 have successfully worked with Foundry ServerIron's, and Vovida's Open Source
 load-balancing proxy, in the past."

Example: SER, Asterisk and Lucent TNT

by Michael Shuler
You can do what we did and setup 2 SER servers which are load balanced by a Foundry ServerIron XL (you could use UltraMonkey for free if you prefer). The 2 SER machines handle the REGISTER messages, NAT and final delivery to the VoIP devices and to the media gateways. The SER machines don't know what to do with a call they only know to hand it over to Asterisk for routing/CLASS features or whatever you want the call to do. You then have a separate set of Asterisk boxes that are nothing but media gateways/transcoders that have the PRI cards in them. We don't actually use Asterisk as media gateways in our setup we use the Lucent TNTs but it accomplishes the same thing but on a much larger scale. As far as sharing the common configs we use http://svn.asteriskdocs.org/res_data/ to read all of our configs (sip.conf, extensions.conf, etc.) live from a MySQL database. Just apply the patch and go.

So far we have seen no limit as to the scalability of this setup. If the 2 SER servers get overloaded we just add another. If the Asterisk routing servers get overloaded we just add another. IF we run out of PRI's we add another TNT and more PRI's from our Plexus 9000 switch. So the system in theory has no real-world limits.

Here is the sample configuration for SER.

Example TeleSIP

Q: Are you aware of any documentation on how to configure SER to be a front-end to Asterisk?

A: At TeleSIP we run a cluster of several geographically distributed SER Servers that hande all our SIP Routing. SER is a robust, fast and stable platform which has worked flawlessly for us. We use * as our company PBX and PSTN Gateway. Basically what you need to do is to devise a numbering plan so that SERs routing logic can forward the call to * when it needs to.

R: Another example can be found in the SER handbook, found on IPtel.org. The
example mentions how to use SER as a frontend to a Cisco PSTN gateway, but also applies to using SER as a frontend to Asterisk.

See also

• Load balancing with DUNDi
• Asterisk SIP not-proxy: Why Asterisk cannot be called "SIP proxy"
• Asterisk config sip.conf: sip.conf example 2 for Asterisk with SER
• Asterisk cmd SIPRedirect: Sending out a 302 temporarily moved with Asterisk 1.2.x
• Asterisk ast_data: Asterisk 1.0.x with SER and MWI integration
• Asterisk MWI: Message Waiting Indication for Asterisk
• Asterisk High Availability Solutions and load balancing
• Asterisk dimensioning: Which hardware is needed to handle x calls?
• Asterisk administration
• file descriptors
• Linux High Availability Project
• IAX load balancer by Stanisław Pitucha

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