22 Nov
2004
22 Nov
'04
10:46 a.m.
Thank you for sharing your views on using ser for carrier-grade services. I
find that open source projects often lack information/sharing on
scalability, redundancy, monitoring and other important things in a
carrier-grade solution. Such information is often considered a competitive
advantage with the result that the open source project looses a lot of the
potential carrier-grade improvements.
I share with you the belief that ser can be put into production for
carrier-grade services. My company has a focus on end-user centric service
delivery and provide managed services, as well as turn-key solutions for
service providers. Our focus so far has been on home/mobile office services
with dial-up, VPN, (two-factor) authentication etc. Our platform has a
powerful and flexible service authentication and authorization engine at the
core, with RADIUS as one front-end (using LDAP backend). In addition, we
have a web front-end for end-user self-service and service administration,
as well as an XML interface for CRM integration. Finally, we have a
Windows-based software that can automatically configure a user's PC, install
components, keep them updated, and also be a "menu" or guide for the user in
using the services (voicemail, forwarding etc).
Our service providers often have multiple agents, channels, resellers.
Each with branding requirements, as well as CRM system integration
requirements. So, several virtual providers can be enabled on the same
platform. Of course, providing billing records is a must...
We are currently adding VoIP as a service option in our platform. I am
sure many people on this list are working on commercial projects each with a
slightly different angle. :-)
So more directly to the perspectives in your email:
- Interesting that you have selected WhiteBox as the distro. Surely, a
RedHat subscription at $349 should not really add a lot to the costs? We
have looked at both CentOs, another RHEL-based distro, and WhiteBox, but
find that the $349 is pretty cheap for the "luxury" of not having to worry
about distros dying or introducing problems in libraries etc.
- You have opted for the development version of ser. We have been
(conservatively) staying with the 0.8.14. Any particular reason for going
for development? (Except for cool features; I think the AVP functionality is
particularly interesting) What is the experienced stability? Will you put
the development version into carrier production?!
- SEMS is a bit immature and I have honestly been in doubt whether I should
spend any time on it at this stage. We have looked at Asterisk as an option
(no time for religious wars even though there is some overlapping
functionality). In fact, we also evaluate PSTN-based voicemail solutions
for scalability and reliability. I must admit, it's a bit backwards, but...
- I'm surprised that you have set up serweb at all. Our experience in
delivering end-user services is that your Achilles heel is support. There
is no limit to what a naive end-user can assume or believe and it will most
definitely cause you a support call.... I look at serweb as an example
application, and I think it would be very interesting to make a LGPL'ed
library for ser functions that could be accessible through soap services.
This way one can easily integrate ser functionality into the gazillion of
CRM and customer front-end systems that you find in the telco world.
- You mention an enum directory for subscribers. I'm pretty sure we're going
there, but there quite some issues related to service provider roaming, how
it is charged etc. Of course, existing telcos are keen to stop any all-IP,
non-billable efforts, and those of us who deliver to telcos must (at least
to a certain degree) adhere to their demands. Non-incumbents will lead the
way here, but I believe that we need to establish a viable business model
for exchanging traffic. The market (or the providers) are not ready for
all-you-can-all-around-the-world options yet. As long as minutes are
charged between IP and PSTN, there will be an incentive to move as much
traffic as possible through PSTN (also for the non-incumbents with such
business models).
Well, talking about rambling... I think the point is: By showing our cards
a bit, do we have some common interests that can benefit the ser community
or create a new business relationship?
g-)
Java Rockx wrote:
Greger,
I believe the small change to the onfailure_route[] was the thing
that made the difference. Andrei pointed out a small error to me
where I was calling fix_nated_contact() for more messages than I
should have. (THANKS ANDREI!!!)
The other thing that made a **substantial** improvement was the fact
that our PSTN gateway provider that uses Sonus equipment as added as
an "alias" at the top of the ser.cfg. This caused ser to not include
"Route:" headers on ACKs and this was causing the Sonus equipment to
drop calls after two minutes.
It's funny how such small changes to ser.cfg can have dramatic
effects.
I'd love to see a complete book published on ser. I'm sure, with the
help of the maintainers and serusers participants this could become a
reality. I mean this is the best damned SIP router I know of.
Commercial solutions such as Sonus are not really a "solution"
because they run as much as US$2 million per box and most small
companies can't afford "one". And even if you could afford =>one<=
you may not be able to afford others for scaling your VoIP platform -
so whats the point of attempting VoIP if you can't scale?
Our solution is modeled after the Google network whereby:
unlimited horizontal scalablity == unlimited cps (calls per second)
And this equates to:
successful VoIP Company == retired employee
If we can indeed do this for our implementation and do it on low end
x86 or Opteron boxes (or blade servers), then the sky is the limit.
Our engineer whom we acquired from RedHat a few months ago already
has a full TCP/IP and UDP socket patch working so we can seperate ser
and serweb and scale them independently of each other (over a VPN)
since we don't need ser's existing FIFO.
SEMS is a whole other beast. We are now working on this using the ivr
module with sipums for a carrier-grade voicemail solution. Early
indications is that we still have plenty to do in order to get
99.999% reliablity with sems. Our goal is to not have sems run on the
same box as ser as we already know this is bad. Voicemail and SIP
proxy cannot be the same box if you expect to be a big VoIP provider,
IMHO.
So our high-level VoIP platform uses the following as independent
server farms:
MySQL 4.1.7 (farm 1)
ser-0.8.99-dev17 (farm 2)
serweb on Apache2 farm 3)
sems/ivr/sipums (farm 4, still a work in progress)
Third party PSTN gateway *** see comment below
All servers use Whitebox Linux Respin 1 and our network
implementation allows us to completely administer all servers at any
remote co-lo site --- even down to remote installation of the
operating system. This includes make a change to the master ser.cfg
and having it replicate to all ser proxies.
Monitoring is a big part of this network architecture and we've only
got basic monitoring right now, so this will become a focus of our
efforts soon.
We also use third parties for providing PSTN access. IMHO, PSTN
gateways are much more trouble than their worth. We'd rather rely on
multiple third parties for this than to try and do it ourselves. The
cost of DSP boards is expensive, especially if you wanted to handle
say 50,000 concurrent callers. For this very reason we use companies
such as Level 3 and Global Crossing and all we have to do is simply
monitor their QoS and route PSTN callers to other PSTN gateways as
needed. And this is all at the network level, so ser is totally
oblivious to this.
Another thing I've love to see is a database whereby VoIP service
providers can lookup 10-digit DIDs in a common database (much like a
root DNS server) and determine if a callee is a VoIP user and route
the SIP call directly from our SIP proxy to their SIP proxy rather
than making a SIP->PSTN->SIP call. We have the Chairman of the IPCC
deeply invovled in our company and he could get the ball rolling on
this, but we don't want to get distracted right now.
Anyhow, I've rambled long enough.
Regards,
Paul Hazlett
We'll publish our
--- "Greger V. Teigre" <greger(a)teigre.com> wrote:
Paul,
Thanks for posting your resulting config after so many hours of
testing! I did a diff and can see that you have done several
(smaller) changes. But what did the trick? (i.e. for Route problem)
g-)
Java Rockx wrote:
=== message truncated ===
__________________________________
Do you Yahoo!?
The all-new My Yahoo! - Get yours free!
http://my.yahoo.com Greger,
I finally got everything working without STUN and without an
Outbound Proxy. I an only using rtpproxy/nathelper.
I've tested these settings with the following SIP Phones/ATAs
Sipura
UTstarcom iAN-02EX
Grandstream ATA486
Grandstream BT100
Cisco ATA186
Cisco 7960G
WorldAccxx ATA
X-Ten Pro
X-Ten Lite
We are very happy with everything now. The only piece of the puzzle
that I don't have working yet is sems/sipums for voicemail - but I'm
working on that.
Attached is my complete ser.cfg file that is working. Please note
that I'm using ser-0.8.99-dev17 for this rather than ser-0.8.14
If anyone finds problems in this ser.cfg then please let me know -
but like I said before, things seem very stable right now with
-dev17.
Regards,
Paul
--- "Greger V. Teigre" <greger(a)teigre.com> wrote:
> Good to get an authoriative answer on this. I think I should
> allocate some time to read the RFC thoroughly.
> This leads back to my guess that it could be Paul's outbound proxy
> setting that fixed the problem. I would think that are some
> Grandstream people this list, but anyway a bug report should be
> submitted to Grandstream. This was a hard bug to track. Paul?
> g-)
>
> Jan Janak wrote:
>> No, ser does not change Record-Route to Route header field, user
>> agents are supposed to do it. SER does only two things:
>>
>> 1) Adds Record-Route header fields with its own IP address (this
>> is what record_route function does)
>> 2) Removes the topmost Route header field if it contains its own
>> IP address (according to the IP) and forwards the message to
>> the IP in the next Route header field if any. If there is no
>> other Route header field then the Request-URI would be used.
>>
>> If there are some Route header fields missing in ACK then this is
>> a bug in the calling user agent, not SER.
>>
>> Jan.
>>
>> On 18-11 21:16, Greger V. Teigre wrote:
>>> I believe the changes are done in the rr module, in the loose.c
>>> file. RFC3261 defines this (as mentioned by the Sonus guys).
>>> I remember vaguely reading something about equivalence between
>>> defining outbund proxy on the client and a Route header, but I'm
>>> way off stable ground here... However, if I remember correctly,
>>> it is probably the outbound proxy and not the stun settings that
>>> does the trick. I have seen some discussions on loose routing
>>> earlier this fall, maybe a search on loose routing in the
>>> archives can turn up some new approaches?
>>>
>>> I'm afraid I don't have anything more to contribute here. From
>>> all I can see, ser should change the Record-Routes to Route, but
>>> doesn't, and I don't understand why. I think we need somebody
>>> with a more in-depth understanding of the ser inner workings.
>>> g-)
>>>
>>>
>>> ----- Original Message -----
>>> From: "Java Rockx" <javarockx(a)yahoo.com>
>>> To: "Greger V. Teigre" <greger(a)teigre.com>om>; "ser
users"
>>> <serusers(a)lists.iptel.org> Sent: Thursday, November 18, 2004 08:21 PM
>>> Subject: Re: [Serusers] Revisted Error: force_rtp_proxy2: can't
>>> extract bodyfrom the message
>>>
>>>
>>>> Greger,
>>>>
>>>> Do you have any idea how SER decides to include a "Route:"
>>>> versus a "Record-Route:" header? If so,
>>>> which piece of code in ser would write the second ACK below?
>>>>
>>>> Here is a "200 OK" and two ACKs - The first ACK is good and
the
>>>> second ACK is bad because it
>>>> should have a "Route:" header referring to the Sonus box.
>>>>
>>>> 100.99.99.99.99 is my SER proxy
>>>> 100.10.10.10 is the public side of my firewall
>>>> 216.50.50.50 is the ip of the Sonus box
>>>>
>>>> So the ACK from SER to Sonus is incorrect.
>>>>
>>>> Do you think this is worth posing to Jiri, Andrei, and company?
>>>> All I know is that this ACK is bad
>>>> when STUN is not used and it is good when STUN is used. I did
>>>> upgrade my Grandstream, but that
>>>> didn't help, and I've modified my nat_uac_test to use mode==19
>>>> rather than mode==3, but still get
>>>> the same results.
>>>>
>>>> Regards,
>>>> Paul
>>>>
>>>> U 2004/11/18 14:13:08.419098 100.99.99.99:5060 ->
>>>> 100.10.10.10:5060 SIP/2.0 200 OK.
>>>> Via: SIP/2.0/UDP
>>>>
192.168.0.83;rport=5060;received=100.10.10.10;branch=z9hG4bKa70081ccdd52daf0.
>>>> To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=069c9797.
>>>> From: "Paul (1002)"
>>>>
<sip:9990010001@sip.mycompany.com;user=phone>;tag=92691bb29380c100.
>>>> Call-ID: e37c04be3e50ea72(a)192.168.0.83.
>>>> CSeq: 21752 INVITE.
>>>> Contact: sip:4075551212@216.50.50.50:5060.
>>>> Record-Route: <sip:216.50.50.50:5060;lr>.
>>>> Record-Route: <sip:100.99.99.99;ftag=92691bb29380c100;lr=on>.
>>>> Accept: multipart/mixed, application/sdp, application/isup,
>>>> application/dtmf,
>>>> application/dtmf-relay.
>>>> Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO.
>>>> Supported: timer.
>>>> Content-Disposition: session;handling=required.
>>>> Content-Type: application/sdp.
>>>> Session-Expires: 240;refresher=uas.
>>>> .
>>>> v=0.
>>>> o=Sonus_UAC 18748 26881 IN IP4 216.229.118.76.
>>>> s=SIP Media Capabilities.
>>>> c=IN IP4 100.99.99.99.
>>>> t=0 0.
>>>> m=audio 35552 RTP/AVP 18 101.
>>>> a=rtpmap:18 G729/8000.
>>>> a=rtpmap:101 telephone-event/8000.
>>>> a=fmtp:101 0-15.
>>>> a=sendrecv.
>>>> a=nortpproxy:yes.
>>>>
>>>> #
>>>> U 2004/11/18 14:13:08.428394 100.10.10.10:5060 ->
>>>> 100.99.99.99:5060 ACK sip:4075551212@216.50.50.50:5060 SIP/2.0.
>>>> Via: SIP/2.0/UDP 192.168.0.83;branch=z9hG4bKf4bb608e498ec61d.
>>>> Route: <sip:100.99.99.99;ftag=92691bb29380c100;lr=on>.
>>>> Route: <sip:216.50.50.50:5060;lr>.
>>>> From: "Paul (1002)"
>>>>
<sip:9990010001@sip.mycompany.com;user=phone>;tag=92691bb29380c100.
>>>> To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=069c9797.
>>>> Contact: <sip:9990010001@192.168.0.83;user=phone>.
>>>> Call-ID: e37c04be3e50ea72(a)192.168.0.83.
>>>> CSeq: 21752 ACK.
>>>> User-Agent: Grandstream BT100 1.0.5.16.
>>>> Max-Forwards: 70.
>>>> Allow:
>>>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. .
>>>>
>>>> #
>>>> U 2004/11/18 14:13:08.429879 100.99.99.99:5060 ->
>>>> 216.50.50.50:5060 ACK sip:216.50.50.50:5060;lr SIP/2.0.
>>>> Via: SIP/2.0/UDP
>>>> 100.99.99.99;branch=z9hG4bK2b35.552edb80cbf475b9be9ae3f9db23f960.0.
>>>> Via: SIP/2.0/UDP
>>>>
192.168.0.83;rport=5060;received=100.10.10.10;branch=z9hG4bKf4bb608e498ec61d.
>>>> From: "Paul (1002)"
>>>>
<sip:9990010001@sip.mycompany.com;user=phone>;tag=92691bb29380c100.
>>>> To: <sip:14075551212@sip.mycompany.com;user=phone>;tag=069c9797.
>>>> Contact: <sip:9990010001@100.10.10.10:5060;user=phone>.
>>>> Call-ID: e37c04be3e50ea72(a)192.168.0.83.
>>>> CSeq: 21752 ACK.
>>>> User-Agent: Grandstream BT100 1.0.5.16.
>>>> Max-Forwards: 16.
>>>> Allow:
>>>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. .
>>>>
>>>>
>>>> --- "Greger V. Teigre" <greger(a)teigre.com> wrote:
>>>>
>>>>> A few comments/questions after having looked at your config
>>>>> file: - You are using 1.0.5.11 firmware on your Grandstream.
>>>>> 1.0.5.16 is the latest and a lot of things have happened since
>>>>> 11. I would suggest you upgrade the firmware before you spend
>>>>> more time on it. - The SIP conversation you sent earlier shows
>>>>> a conversation without using
>>>>> STUN. In later messages you write about setting outbound and
>>>>> then stun. Have you tried setting outbound, but not stun? Any
>>>>> difference in Route?
>>>>> - From 1.0.5.7, Grandstream started to send stun-detected ip
>>>>> and port when
>>>>> symmetric NAT was found. My experience is that if stun is set
>>>>> on