siproxd-users — Users Mailing List

[Siproxd-users] registration and header

[Franck B. <fr...@fb...>]

Hello,
I use siproxd 0.5.13 and lib64osip2 2.2.2 on a mandriva spring ( 2007.1 ).
Then i try to register to the sip server ( asterisk ) with Express Talk from inside my lan ( 172.16.0.44 ) , i've got this log :

 siproxd: sock.c:125 received UDP packet from 172.16.0.44, count=399
 siproxd: accessctl.c:53 deny  list (SIP):*NULL*
 siproxd: accessctl.c:55 allow list (SIP):*NULL*
 siproxd: accessctl.c:57 allow list (REG):172.16.0.0/16
 siproxd: accessctl.c:154 [0] extracted address=172.16.0.0
 siproxd: accessctl.c:155 [0] extracted mask   =16
 siproxd: utils.c:114 DNS lookup - from cache: 172.16.0.0 -> 172.16.0.0
 siproxd: accessctl.c:172 check match: entry=0, filter=ac100000, from=ac100000
 siproxd: accessctl.c:95 granted REG/SIP access
 siproxd: accessctl.c:102 access check =3
 siproxd: security.c:48 security_check_raw: size=399
 siproxd: security.c:268 ERROR:security check failed: NULL To Header
 siproxd: siproxd.c:348 ERROR:security_check_sip() failed... this is not good
 siproxd: siproxd.c:270 going into sipsock_wait 

I've try with x-lite too, and it's the same result : NULL To Header
What can i do to solve this ?
Thank's in advance

Franck Barel
fr...@fb...

Re: [Siproxd-users] ERROR:I'm trying to delete a VIA but it's not mine!

[Colin G. <gm...@co...>]

Colin Guthrie wrote:
> Thomas Ries wrote:
>> Hmm, it's now yet completely clear to me how your actual setup looks
>> like. Why do you use 
>>
>> if_inbound  = eth1
>> if_outbound = eth1
>>
>> ? Do you really have your local UA's *and* the conenction to the outside
>> world on the same interface (logical AND physical)?
> 
> Erm, (colin potentially get's embarrased), no.... there are two
> interfaces, WAN and local, eth0 is WAN and eth1 is local/internal.

Right, just some feedback.

I've now configured all the phones in the office to route through
siproxd and made the necessary config changes.

Since last night all has worked flawlessly and I have a 130Meg debug log!

No resets needed yet. I'll leave it over the weekend and hopefully all
will be well come Monday (I'll post a follow up just for completeness)

Thanks loads for all your help. Sorry it seems it was mostly down to
idiocy that certain things were not working :p

Cheers

Col

Re: [Siproxd-users] ERROR:I'm trying to delete a VIA but it's not mine!

[Colin G. <gm...@co...>]

Thomas Ries wrote:
> Hmm, it's now yet completely clear to me how your actual setup looks
> like. Why do you use 
> 
> if_inbound  = eth1
> if_outbound = eth1
> 
> ? Do you really have your local UA's *and* the conenction to the outside
> world on the same interface (logical AND physical)?

Erm, (colin potentially get's embarrased), no.... there are two
interfaces, WAN and local, eth0 is WAN and eth1 is local/internal.

>From this comment in siproxd.conf:
#    If siproxd is not running on the host doing the masquerading
#    but on a host within the private network segment, "in front" of
#    the masquerading router: define if_inbound and if_outbound to
#    point to the same interface (the inbound interface). In *addition*
#    define 'host_outbound' to hold your external (public) IP address
#    or a hostname that resolves to that address (use a dyndns address
#    for example).

I thought it was correct to set them both to eth1 (local) as this is
"inbound" for the local SIP clients (obviously it is also inbound for
the externally initiated calls too...).

Also, in my case my server is "in between" the masquerading router,
rather than "in front".

e.g. is the comment above referring to:


                 +---------+
                 | siproxd |
                 +---------+
+------------+        |        +-------------+
| SIP Client |--------+--------| Masq Router |------- INTERWEB
+------------+                 +-------------+


Where as my setup is more like:

+------------+   +---------+   +-------------+
| SIP Client |---| siproxd |---| Masq Router |------- INTERWEB
+------------+   +---------+   +-------------+
                 eth1   eth0   Int IP     Public IP

Where:
 eth1 = 192.168.112.1
 eth0 = 192.168.0.1
 Int IP (of the router) is 192.168.0.254
 Public IP is 84.9.255.116

So I'll try with:

if_inbound = eth1 (local)
if_outbound = eth0 (WAN)

to see if that helps.

> I understand you in a way that the interface eth0 owns the IP address
> 192.168.0.1, so this IP is NOT seen as an "own" IP for siproxd, as the
> eth0 interface is not mentioned in its configuration. So make sure your
> internal UAs do use the proper internal IP address (the one where the
> if_inbound points to).

I think I covered this above.


> If you plan to deal with multiple internal subnets (without routing in
> between them) then you need a siproxd instance for each subnet (put them
> on different UDP ports, e.g 5060, 5062).

The only separate subnet is the one to the router (this is also the
wireless subnet too, but I don't care about that right now - wireless is
considered external for all intent purposes).

> However, I still would like to get a debug log file (debuglevel=-1) of
> such an event. As I understood you correctly, your siproxd machine uses
> 192.111.200.139 on eth1 and 192.168.0.1 on eth0. So it is not clear why
> both IPs are involved...

I'll turn this on now (debug file).

The IP 192.111.200.139 is that of my SIP provider
(lon-pbx-3.gradwell.net). My public IP is given above.

I'll try again with inbound/outbound more sensibly set (should have
thought about it more rather than blindly following the comment!)

> And yes, siproxd requires a DNS environment that is working. It is not
> the only component on a SIP network that relies on it. If the DNS from
> your provider / SIP registrar is not that reliable, you could:
> 
> - introduce a caching nameserver for your local network
> - use the plain IP address in siproxd config file

Yeah the weird thing is we have a caching nameserver! Go figure! I've
added the DNS to /etc/hosts just now.

Thanks again for all your help! I really appreciate it! I'm getting a
good slagging off in our office for insisting we go VoIP and the
reliability so for has been rubbish. It's getting better now I've
introduced siproxd so you are a legend in my eyes!

Col.

Col

Re: [Siproxd-users] ERROR:I'm trying to delete a VIA but it's not mine!

[Thomas R. <tr...@gm...>]

Hmm, it's now yet completely clear to me how your actual setup looks
like. Why do you use 

if_inbound  = eth1
if_outbound = eth1

? Do you really have your local UA's *and* the conenction to the outside
world on the same interface (logical AND physical)?
I understand you in a way that the interface eth0 owns the IP address
192.168.0.1, so this IP is NOT seen as an "own" IP for siproxd, as the
eth0 interface is not mentioned in its configuration. So make sure your
internal UAs do use the proper internal IP address (the one where the
if_inbound points to).

If you plan to deal with multiple internal subnets (without routing in
between them) then you need a siproxd instance for each subnet (put them
on different UDP ports, e.g 5060, 5062).

However, I still would like to get a debug log file (debuglevel=-1) of
such an event. As I understood you correctly, your siproxd machine uses
192.111.200.139 on eth1 and 192.168.0.1 on eth0. So it is not clear why
both IPs are involved...


And yes, siproxd requires a DNS environment that is working. It is not
the only component on a SIP network that relies on it. If the DNS from
your provider / SIP registrar is not that reliable, you could:

- introduce a caching nameserver for your local network
- use the plain IP address in siproxd config file

/Thomas

On 26 Jun, Colin Guthrie wrote:
> Colin Guthrie wrote:
>> Thomas Ries wrote:
>>> Hi Colin,
>>>
>>> If you still get the error about deleting vias (and you are sure
>>> that the via in question is your external IP address), please send
>>> me a debug log (loglevel = -1) including the error (and all output
>>> before, starting when the offending DIALOG started).
> 
>> Thanks for all the info. I'll look at this on Monday when I'm in the
>> office!
> 
> OK here is some more info.
> 
> (no debug log yet tho but perhaps you can infer from this?) I'll start
> logging ASAP.
> 
> Reminder (from siproxd.conf):
> if_inbound  = eth1
> if_outbound = eth1
> host_outbound = 84.9.255.116
> outbound_domain_name = lon-pbx-3.gradwell.net
> outbound_domain_host = nat1.gradwell.net
> outbound_domain_port = 5082
> 
> 
> Looks like a DNS error in looking up nat1.gradwell.net caused an issue:
> 
> Jun 26 10:24:00 summit siproxd[20879]: utils.c:185
> ERROR:gethostbyname(nat1.gradwell.net) failed: h_errno=2 [Host name
> lookup failure]
> Jun 26 10:24:00 summit siproxd[20879]: sip_utils.c:916
> ERROR:sip_find_outbound_proxy: cannot resolve outbound proxy host
> [nat1.gradwell.net], check config
> Jun 26 10:24:03 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm
> trying to delete a VIA but it's not mine! host=193.111.200.139
> Jun 26 10:24:03 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm
> trying to delete a VIA but it's not mine! host=193.111.200.139
> Jun 26 10:24:03 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm
> trying to delete a VIA but it's not mine! host=192.168.0.1
> Jun 26 10:24:22 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm
> trying to delete a VIA but it's not mine! host=192.168.0.1
> Jun 26 10:24:53 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm
> trying to delete a VIA but it's not mine! host=192.168.0.1
> Jun 26 10:29:55 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm
> trying to delete a VIA but it's not mine! host=193.111.200.139
> 
> 
> This appears to have caused knock on effects!
> 
> The IP 192.111.200.139 is lon-pbx-3.gradwell.net and 192.168.0.1 is
> the internal IP the server uses when connecting to the wider interweb
> (e.g. eth0). Perhaps siproxd needs to know about eth0 in some way too?
> Or perhaps this isn't an issue if the 193.111.200.139 "delete VIA"
> error is not triggered.
> 
> I'll setup logging now, but perhaps this will help.
> 
> I'm tempted to add nat1.gradwell.net and lon-pbx-3.gradwell.net into
> my
> /etc/hosts file, but perhaps there should be a more robust DNS cache
> in siproxd?
> 
> Col
> 
> 
> -------------------------------------------------------------------------
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> Download DB2 Express C - the FREE version of DB2 express and take
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Re: [Siproxd-users] ERROR:I'm trying to delete a VIA but it's not mine!

[Colin G. <gm...@co...>]

Colin Guthrie wrote:
> I'm tempted to add nat1.gradwell.net and lon-pbx-3.gradwell.net into my
> /etc/hosts file, but perhaps there should be a more robust DNS cache in
> siproxd?

FWIW, adding these addresses to /etc/hosts didn't make any difference.


Col

Re: [Siproxd-users] ERROR:I'm trying to delete a VIA but it's not mine!

[Colin G. <gm...@co...>]

Colin Guthrie wrote:
> Thomas Ries wrote:
>> Hi Colin,
>>
>> If you still get the error about deleting vias (and you are sure that
>> the via in question is your external IP address), please send me a debug
>> log (loglevel = -1) including the error (and all output before, starting
>> when the offending DIALOG started).

> Thanks for all the info. I'll look at this on Monday when I'm in the office!

OK here is some more info.

(no debug log yet tho but perhaps you can infer from this?) I'll start
logging ASAP.

Reminder (from siproxd.conf):
if_inbound  = eth1
if_outbound = eth1
host_outbound = 84.9.255.116
outbound_domain_name = lon-pbx-3.gradwell.net
outbound_domain_host = nat1.gradwell.net
outbound_domain_port = 5082


Looks like a DNS error in looking up nat1.gradwell.net caused an issue:

Jun 26 10:24:00 summit siproxd[20879]: utils.c:185
ERROR:gethostbyname(nat1.gradwell.net) failed: h_errno=2 [Host name
lookup failure]
Jun 26 10:24:00 summit siproxd[20879]: sip_utils.c:916
ERROR:sip_find_outbound_proxy: cannot resolve outbound proxy host
[nat1.gradwell.net], check config
Jun 26 10:24:03 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm trying
to delete a VIA but it's not mine! host=193.111.200.139
Jun 26 10:24:03 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm trying
to delete a VIA but it's not mine! host=193.111.200.139
Jun 26 10:24:03 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm trying
to delete a VIA but it's not mine! host=192.168.0.1
Jun 26 10:24:22 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm trying
to delete a VIA but it's not mine! host=192.168.0.1
Jun 26 10:24:53 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm trying
to delete a VIA but it's not mine! host=192.168.0.1
Jun 26 10:29:55 summit siproxd[20879]: sip_utils.c:620 ERROR:I'm trying
to delete a VIA but it's not mine! host=193.111.200.139


This appears to have caused knock on effects!

The IP 192.111.200.139 is lon-pbx-3.gradwell.net and 192.168.0.1 is the
internal IP the server uses when connecting to the wider interweb (e.g.
eth0). Perhaps siproxd needs to know about eth0 in some way too? Or
perhaps this isn't an issue if the 193.111.200.139 "delete VIA" error is
not triggered.

I'll setup logging now, but perhaps this will help.

I'm tempted to add nat1.gradwell.net and lon-pbx-3.gradwell.net into my
/etc/hosts file, but perhaps there should be a more robust DNS cache in
siproxd?

Col

Re: [Siproxd-users] ERROR:I'm trying to delete a VIA but it's not mine!

[Colin G. <gm...@co...>]

Thomas Ries wrote:
> Hi Colin,
> 
> If you still get the error about deleting vias (and you are sure that
> the via in question is your external IP address), please send me a debug
> log (loglevel = -1) including the error (and all output before, starting
> when the offending DIALOG started).
> 
> An error of "unable to parse SIP message" does mean that the libosip2
> parser did fail to parse the SIP message. Usually this is due to
> incorrect SIP headers (on classic is having Asterisk adding an
> Alert-Info header without using the '<>' around the value part).
> What you need is the SIP message that provokes this error. That message
> will be included in the debug output for any debuglevel != 0. Then you
> have to investigate header for header.
> I usually first create a simple text file containing the exact UDP
> playload of the offending SIP packet and send it to siproxd using
> netcat. This must provoke the error. Then, I start removing / modifying
> the headers until I get the guilty. RFC3261 does include a detailed
> syntax description of how the headers must look like, which are
> mandatory and which are optional.

Thanks for all the info. I'll look at this on Monday when I'm in the office!

Thanks loads.

Col

Re: [Siproxd-users] ERROR:I'm trying to delete a VIA but it's not mine!

[Thomas R. <tr...@gm...>]

Hi Colin,

If you still get the error about deleting vias (and you are sure that
the via in question is your external IP address), please send me a debug
log (loglevel = -1) including the error (and all output before, starting
when the offending DIALOG started).

An error of "unable to parse SIP message" does mean that the libosip2
parser did fail to parse the SIP message. Usually this is due to
incorrect SIP headers (on classic is having Asterisk adding an
Alert-Info header without using the '<>' around the value part).
What you need is the SIP message that provokes this error. That message
will be included in the debug output for any debuglevel != 0. Then you
have to investigate header for header.
I usually first create a simple text file containing the exact UDP
playload of the offending SIP packet and send it to siproxd using
netcat. This must provoke the error. Then, I start removing / modifying
the headers until I get the guilty. RFC3261 does include a detailed
syntax description of how the headers must look like, which are
mandatory and which are optional.

Regards,
/Thomas

On 24 Jun, Colin Guthrie wrote:
> Hi Thomas,
> 
> Thomas Ries wrote:
>> Two things.
>> First, the ERROR you describe about trying to delete a VIA that is
>> not mine: There was a bug that did take the "real" external IP
>> address (host_outbound config directive) to be a siproxd local
>> address - resulting in this error (usually during incoming responses
>> only).
> 
>> The bug (issue #1) is fixed, you may want to try the latest snapshot,
>> I hope the ERRORs should now be gone completely - and maybe also the
>> need to reset the phones.
> 
> Well I've updated to the latest snapshot but unf. these errors are
> still occurring for me. I've not reset the phones (I'm not actually in
> the office, so perhaps this is a hangover from not clearing out the
> phones own internal caches but (perhaps naively) I presumed the bug
> would be localised to the proxy only.
> 
> 
>> Second, you say you need to use another (chained) proxy to get things
>> work. This may be perfectly legal - some providers require this
>> (possibly to overcome firewall/NAT issues on their own side). Not
>> using this additional proxy will result in "things almost working".
>> However, this should be somehow documented on your providers side
>> (configuration help / examples that show you requires their proxy).
> 
> The provider unfortunately doesn't support the use of a local proxy
> e.g. siproxd, so we're the first client to use it!
> 
> I'll maybe try doing a few trial and error things on Monday when I'm
> back in the office. Pehraps with the latest snapshot and the VIA
> errors being fixed I can now remove the proxy, so we'll see if that
> works now :)
> 
> 
> 
> As a quick (unrelated) question, if I get an error about being upable
> to parse SIP message, what level of debug would be useful to you? My
> SNOM 360 seems to get hit by this problem if I channel it through the
> proxy, but no point in me reporting anything unless I can give you
> useful debug :)
> 
> 
> Cheers.
> 
> Col

Re: [Siproxd-users] ERROR:I'm trying to delete a VIA but it's not mine!

[Colin G. <gm...@co...>]

Hi Thomas,

Thomas Ries wrote:
> Two things.
> First, the ERROR you describe about trying to delete a VIA that is not
> mine: There was a bug that did take the "real" external IP address
> (host_outbound config directive) to be a siproxd local address -
> resulting in this error (usually during incoming responses only).

> The bug (issue #1) is fixed, you may want to try the latest snapshot,
> I hope the ERRORs should now be gone completely - and maybe also the
> need to reset the phones.

Well I've updated to the latest snapshot but unf. these errors are still
occurring for me. I've not reset the phones (I'm not actually in the
office, so perhaps this is a hangover from not clearing out the phones
own internal caches but (perhaps naively) I presumed the bug would be
localised to the proxy only.


> Second, you say you need to use another (chained) proxy to get things
> work. This may be perfectly legal - some providers require this
> (possibly to overcome firewall/NAT issues on their own side). Not using
> this additional proxy will result in "things almost working". However,
> this should be somehow documented on your providers side (configuration
> help / examples that show you requires their proxy).

The provider unfortunately doesn't support the use of a local proxy e.g.
siproxd, so we're the first client to use it!

I'll maybe try doing a few trial and error things on Monday when I'm
back in the office. Pehraps with the latest snapshot and the VIA errors
being fixed I can now remove the proxy, so we'll see if that works now :)



As a quick (unrelated) question, if I get an error about being upable to
parse SIP message, what level of debug would be useful to you? My SNOM
360 seems to get hit by this problem if I channel it through the proxy,
but no point in me reporting anything unless I can give you useful debug :)


Cheers.

Col

Re: [Siproxd-users] ERROR:I'm trying to delete a VIA but it's not mine!

[Thomas R. <tr...@gm...>]

Hi Col,

Two things.
First, the ERROR you describe about trying to delete a VIA that is not
mine: There was a bug that did take the "real" external IP address
(host_outbound config directive) to be a siproxd local address -
resulting in this error (usually during incoming responses only).

Second, you say you need to use another (chained) proxy to get things
work. This may be perfectly legal - some providers require this
(possibly to overcome firewall/NAT issues on their own side). Not using
this additional proxy will result in "things almost working". However,
this should be somehow documented on your providers side (configuration
help / examples that show you requires their proxy).

The bug (issue #1) is fixed, you may want to try the latest snapshot,
I hope the ERRORs should now be gone completely - and maybe also the
need to reset the phones.


Regards,
/Thomas

On 21 Jun, Colin Guthrie wrote:
> Hi,
> 
> I've just starting using siproxd as it appears to be the ideal
> solution to my network setup.
> 
> We have a small office and are behind a NAT router and want to use
> several hardware phones and Ekiga in our office.
> 
> Everything is routed through our main office server which runs siproxd
> but it does not do the NAT, this is left to our ADSL Modem which also
> DNATs ALL traffic back to this server for incoming connections.
> 
> On this server the external network is connected to via eth0 and the
> internal via eth1. The external subnet (to talk to the router) is
> 192.168.0.0/24 and the internal is 192.168.112.0/24.
> 
> My /etc/siproxd.conf is as below.
> 
> Through trial and error, I have found that I need to connect via an
> outgoing proxy in order to make things work. Not doing so will give
> the above errors (as per the subject). Unfortunately, recently I've
> found that even doing so still appears to log the occasional error
> about deleting a VIA that is not mine, even tho' the full error lists
> the IP address specified in host_outbound. I would have through that
> siproxd should consider the host_outbound IP to be "one of it's own!"?
> Also I found I needed the three part outbound proxy for things to work
> reliably. This was also through trial and error. I would have thought
> that no proxy should be needed here and that I SHOULD be able to
> connect siproxd directly to the internet due to the fact that all
> incoming connections are DNAT'ed back to the server from the firewall?
> Or is there perhaps a problem in that the outgoing packets from the
> server are NATed by the router? If so should I try to disable this?
> (e.g. move the NATing to the server?) From what I've read this does
> not appear to be a problem.
> 
> The phones in question are not setup to use STUN or any other NAT
> technique. They are just configured to use an outgoing proxy which
> points to the internal machine.
> 
> I have another problem relating to a message "Unable to Parse SIP
> message. This is not good." but I'll write about that in another mail.
> If someone could say what they would need from a debug log for helping
> to fix this I could make sure this is done correctly first post. Also
> any advice as to what to blank out would help!
> 
> Generally I find that even when things work for the most part, I will
> have to reset all the phones once a day which is very annoying!
> 
> I am using the latest snapshot build of siproxd.
> 
> FWIW I'm connecting to Gradwell's VOIP service.
> 
> I know this list does not have much traffic so if there is a better
> way to request some help then please let me know!
> 
> Thanks.
> 
> Col
> 
> 
> 
> if_inbound  = eth1
> if_outbound = eth1
> host_outbound = <OUR EXTERNAL STATIC IP>
> hosts_allow_reg = 192.168.0.0/16
> sip_listen_port = 5060
> daemonize = 1
> silence_log = 1
> log_calls = 1
> user = siproxd
> registration_file = /var/lib/siproxd/siproxd_registrations
> autosave_registrations = 300
> pid_file = /var/run/siproxd/siproxd.pid
> rtp_proxy_enable = 1
> rtp_port_low  = 7070
> rtp_port_high = 7089
> rtp_timeout = 300
> rtp_dscp = 46
> rtp_input_dejitter  = 100000
> rtp_output_dejitter = 100000
> default_expires = 600
> debug_level =      0x00000004
> debug_port = 0
> outbound_domain_name = lon-pbx-3.gradwell.net
> outbound_domain_host = nat1.gradwell.net
> outbound_domain_port = 5082
> pi_shortdial_enable = 0
> pi_shortdial_akey = *00
> pi_shortdial_entry = 17474743246
> pi_shortdial_entry = 17474745000
> 
> 
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> Download DB2 Express C - the FREE version of DB2 express and take
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GE d s+: a C+++ UL+++ P+++ L++++ E-- W++ N++ o+ K w-- 
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G e++ h r+++ y+++
------END GEEK CODE BLOCK------

[Siproxd-users] ERROR:I'm trying to delete a VIA but it's not mine!

[Colin G. <gm...@co...>]

Hi,

I've just starting using siproxd as it appears to be the ideal solution
to my network setup.

We have a small office and are behind a NAT router and want to use
several hardware phones and Ekiga in our office.

Everything is routed through our main office server which runs siproxd
but it does not do the NAT, this is left to our ADSL Modem which also
DNATs ALL traffic back to this server for incoming connections.

On this server the external network is connected to via eth0 and the
internal via eth1. The external subnet (to talk to the router) is
192.168.0.0/24 and the internal is 192.168.112.0/24.

My /etc/siproxd.conf is as below.

Through trial and error, I have found that I need to connect via an
outgoing proxy in order to make things work. Not doing so will give the
above errors (as per the subject). Unfortunately, recently I've found
that even doing so still appears to log the occasional error about
deleting a VIA that is not mine, even tho' the full error lists the IP
address specified in host_outbound. I would have through that siproxd
should consider the host_outbound IP to be "one of it's own!"? Also I
found I needed the three part outbound proxy for things to work
reliably. This was also through trial and error. I would have thought
that no proxy should be needed here and that I SHOULD be able to connect
siproxd directly to the internet due to the fact that all incoming
connections are DNAT'ed back to the server from the firewall? Or is
there perhaps a problem in that the outgoing packets from the server are
NATed by the router? If so should I try to disable this? (e.g. move the
NATing to the server?) From what I've read this does not appear to be a
problem.

The phones in question are not setup to use STUN or any other NAT
technique. They are just configured to use an outgoing proxy which
points to the internal machine.

I have another problem relating to a message "Unable to Parse SIP
message. This is not good." but I'll write about that in another mail.
If someone could say what they would need from a debug log for helping
to fix this I could make sure this is done correctly first post. Also
any advice as to what to blank out would help!

Generally I find that even when things work for the most part, I will
have to reset all the phones once a day which is very annoying!

I am using the latest snapshot build of siproxd.

FWIW I'm connecting to Gradwell's VOIP service.

I know this list does not have much traffic so if there is a better way
to request some help then please let me know!

Thanks.

Col



if_inbound  = eth1
if_outbound = eth1
host_outbound = <OUR EXTERNAL STATIC IP>
hosts_allow_reg = 192.168.0.0/16
sip_listen_port = 5060
daemonize = 1
silence_log = 1
log_calls = 1
user = siproxd
registration_file = /var/lib/siproxd/siproxd_registrations
autosave_registrations = 300
pid_file = /var/run/siproxd/siproxd.pid
rtp_proxy_enable = 1
rtp_port_low  = 7070
rtp_port_high = 7089
rtp_timeout = 300
rtp_dscp = 46
rtp_input_dejitter  = 100000
rtp_output_dejitter = 100000
default_expires = 600
debug_level =      0x00000004
debug_port = 0
outbound_domain_name = lon-pbx-3.gradwell.net
outbound_domain_host = nat1.gradwell.net
outbound_domain_port = 5082
pi_shortdial_enable = 0
pi_shortdial_akey = *00
pi_shortdial_entry = 17474743246
pi_shortdial_entry = 17474745000

Re: [Siproxd-users] Error "security check failed: NULL To Header" when running siproxd

[Thomas R. <tr...@gm...>]

Hi,

I', currently in vacation, so just a few short comments, see below:

Michael Engel wrote:
> Hi,
>=20
> we're trying to run siproxd as proxy for an Asterisk server running
> behind a NAT in a private IP range. Asterisk runs on a Linux
> (Eisfair) machine, while the NAT router running siproxd is a
> NetBSD Sparc64 machine. siproxd (0.5.13 as well as 0.6.0) and
> libosip2-2.2.2 compiled without problems.
>=20
> The NAT router connects the internal network (192.168.1.x,
> interface hme0) to the Internet (DSL using pppoe with dynamic
> addresses and dyndns service, interface pppoe0). Port 5060
> (outgoing) is redirected on the router using the rule
>=20
> rdr hme0 0/0 port 5060 -> 127.0.0.1 port 5060 tcp/udp
>=20
> in /etc/ipnat.conf, incoming ports 5060 as well as
> 10000-20000 are redirected to the Asterisk server:

Why do you redirect incoming ports? By redirecting them you "steal" them
from siproxd, so they can not be processed. This will not work properly.
As shown in
<http://siproxd.sourceforge.net/siproxd_guide/siproxd_guide_c6s5.html>,
OUTGOING traffic must be passed via siproxd, the incoming traffic is
directed to siproxd and MUST be processed by siproxd. It then will be
sent (by siproxd) to the internal UAs.
The only firewall rules for INCOMING traffic are to ALLOW the incoming
traffic to siproxd.


>=20
> rdr pppoe0 0/0 port 5060-5065 -> 192.168.1.200 port 5060 tcp/udp
> rdr pppoe0 0/0 port 10000-20000 -> 192.168.1.200 port 10000 tcp/udp
>=20
> Now, whenever the Asterisk server tries to register with a
> provider (in the example below it's T-Online, but sipgate and
> gmx cause the same problem), I get an error message:
>=20
> security check failed: NULL To Header
>=20
> This happens with siproxd versions 0.5.13 and 0.6.0. As far
> as I can see from the packet log, there is a valid "To" header
> field, so I suspect s.th. else may fail here (64 bit perhaps?
> Big endian byte order? gcc is version 3.3.3)... any ideas?

This error indicates that the parsed SIP message (osip_message_t
structure) has no valid To header (NULL). At least the host part must be
present.
   ...snip...
   /* check for existing To: header */
   if ((sip->to=3D=3DNULL)||
       (sip->to->url=3D=3DNULL)||(sip->to->url->host=3D=3DNULL)) {
      ERROR("security check failed: NULL To Header");
      return STS_FAILURE;
   }
   ...snip...

It would be very interesting if you could edit the file security.c and
put additional debug output in to see what actually is contained in this
structure. It *might* even be a libosip2 error (I'm not yet going to
blame anyone, but these checks are made right after libosip2 did parse
the received message - and if there is NULL...)




>=20
> I've attached the log excerpt and the config file below, if you
> require more information please let me know...
>=20
> Btw., we're mainly setting this up because gmx VoIP doesn't
> seem to work with the NATed configuration (T-Online and
> sipgate actually work for incoming as well as outgoing calls
> without using a proxy). Do you have any experience running
> siproxd against gmx?
>=20
> -- Michael
>=20
> 02:19:01 ERROR:security.c:268 security check failed: NULL To Header
> 02:19:01 ERROR:siproxd.c:348 security_check_sip() failed... this is =20
> not good
> ---BUFFER DUMP follows---
>    52 45 47 49 53 54 45 52 20 73 69 70 3a 74 65 6c REGISTER sip:tel
>    2e 74 2d 6f 6e 6c 69 6e 65 2e 64 65 20 53 49 50 .t-online.de SIP
>    2f 32 2e 30 0d 0a 56 69 61 3a 20 53 49 50 2f 32 /2.0..Via: SIP/2
>    2e 30 2f 55 44 50 20 38 34 2e 31 36 35 2e xx xx .0/UDP 84.165.xx
>    2e xx xx xx 3a 35 30 36 30 3b 62 72 61 6e 63 68 .xxx:5060;branch
>    3d 7a 39 68 47 34 62 4b 35 64 64 63 61 38 37 62 =3Dz9hG4bK5ddca87b
>    3b 72 70 6f 72 74 0d 0a 46 72 6f 6d 3a 20 3c 73 ;rport..From: <s
>    69 70 3a 30 33 32 32 32 39 32 xx xx xx xx xx xx ip:032229xxxxxx@
>    74 65 6c 2e 74 2d 6f 6e 6c 69 6e 65 2e 64 65 3e tel.t-online.de>
>    3b 74 61 67 3d 61 73 31 39 62 62 63 38 33 36 0d ;tag=3Das19bbc836.
>    0a 54 6f 3a 20 3c 73 69 70 3a 30 33 32 32 32 39 .To: <sip:032229
>    xx xx xx xx xx xx 40 74 65 6c 2e 74 2d 6f 6e 6c xxxxxx@tel.t-onl
>    69 6e 65 2e 64 65 3e 0d 0a 43 61 6c 6c 2d 49 44 ine.de>..Call-ID
>    3a 20 37 66 63 36 63 62 66 32 34 30 34 35 62 62 : 7fc6cbf24045bb
>    33 37 30 31 36 66 63 66 30 38 33 65 64 39 33 66 37016fcf083ed93f
>    62 32 40 31 39 32 2e 31 36 38 2e 31 2e 32 30 30 b2@192.168.1.200
>    0d 0a 43 53 65 71 3a 20 31 30 36 20 52 45 47 49 ..CSeq: 106 REGI
>    53 54 45 52 0d 0a 55 73 65 72 2d 41 67 65 6e 74 STER..User-Agent
>    3a 20 41 73 74 65 72 69 73 6b 20 50 42 58 0d 0a : Asterisk PBX..
>    4d 61 78 2d 46 6f 72 77 61 72 64 73 3a 20 37 30 Max-Forwards: 70
>    0d 0a 45 78 70 69 72 65 73 3a 20 33 36 30 30 0d ..Expires: 3600.
>    0a 43 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 30 .Contact: <sip:0
>    33 32 32 32 39 32 39 33 39 31 33 40 38 34 2e 31 32229293913@84.1
>    36 35 2e xx xx 2e xx xx xx 3e 0d 0a 45 76 65 6e 65.xx.xxx>..Even
>    74 3a 20 72 65 67 69 73 74 72 61 74 69 6f 6e 0d t: registration.
>    0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e 67 74 68 3a .Content-Length:
>    20 30 0d 0a 0d 0a                                0....
>=20
> ---end of BUFFER DUMP---
>=20
> config:
>=20
> if_inbound  =3D hme0
> if_outbound =3D pppoe0
> hosts_allow_reg =3D 192.168.1.0/24
> hosts_allow_sip =3D 192.168.1.0/24

This host_allow_sip line wil prohibit ANYONE not in the 192.168.1.x net
to send anything to siproxd. Does not really make sense (you want to
receive SIP packets from the Internet, I assume?). Comment this out.


> sip_listen_port =3D 5060
> daemonize =3D 0
> silence_log =3D 0
> log_calls =3D 1
> user =3D nobody
> registration_file =3D /var/run/siproxd/siproxd_registrations
> pid_file =3D /var/run/siproxd/siproxd.pid
> rtp_proxy_enable =3D 1
> rtp_port_low  =3D 10000
> rtp_port_high =3D 20000
> rtp_timeout =3D 300
> default_expires =3D 600
> debug_level =3D 0xffffffff
> debug_port =3D 0
>=20


Regards,
/Thomas

[Siproxd-users] Error "security check failed: NULL To Header" when running siproxd

[Michael E. <en...@fa...>]

Hi,

we're trying to run siproxd as proxy for an Asterisk server running
behind a NAT in a private IP range. Asterisk runs on a Linux
(Eisfair) machine, while the NAT router running siproxd is a
NetBSD Sparc64 machine. siproxd (0.5.13 as well as 0.6.0) and
libosip2-2.2.2 compiled without problems.

The NAT router connects the internal network (192.168.1.x,
interface hme0) to the Internet (DSL using pppoe with dynamic
addresses and dyndns service, interface pppoe0). Port 5060
(outgoing) is redirected on the router using the rule

rdr hme0 0/0 port 5060 -> 127.0.0.1 port 5060 tcp/udp

in /etc/ipnat.conf, incoming ports 5060 as well as
10000-20000 are redirected to the Asterisk server:

rdr pppoe0 0/0 port 5060-5065 -> 192.168.1.200 port 5060 tcp/udp
rdr pppoe0 0/0 port 10000-20000 -> 192.168.1.200 port 10000 tcp/udp

Now, whenever the Asterisk server tries to register with a
provider (in the example below it's T-Online, but sipgate and
gmx cause the same problem), I get an error message:

security check failed: NULL To Header

This happens with siproxd versions 0.5.13 and 0.6.0. As far
as I can see from the packet log, there is a valid "To" header
field, so I suspect s.th. else may fail here (64 bit perhaps?
Big endian byte order? gcc is version 3.3.3)... any ideas?

I've attached the log excerpt and the config file below, if you
require more information please let me know...

Btw., we're mainly setting this up because gmx VoIP doesn't
seem to work with the NATed configuration (T-Online and
sipgate actually work for incoming as well as outgoing calls
without using a proxy). Do you have any experience running
siproxd against gmx?

-- Michael

02:19:01 ERROR:security.c:268 security check failed: NULL To Header
02:19:01 ERROR:siproxd.c:348 security_check_sip() failed... this is  
not good
---BUFFER DUMP follows---
   52 45 47 49 53 54 45 52 20 73 69 70 3a 74 65 6c REGISTER sip:tel
   2e 74 2d 6f 6e 6c 69 6e 65 2e 64 65 20 53 49 50 .t-online.de SIP
   2f 32 2e 30 0d 0a 56 69 61 3a 20 53 49 50 2f 32 /2.0..Via: SIP/2
   2e 30 2f 55 44 50 20 38 34 2e 31 36 35 2e xx xx .0/UDP 84.165.xx
   2e xx xx xx 3a 35 30 36 30 3b 62 72 61 6e 63 68 .xxx:5060;branch
   3d 7a 39 68 47 34 62 4b 35 64 64 63 61 38 37 62 =z9hG4bK5ddca87b
   3b 72 70 6f 72 74 0d 0a 46 72 6f 6d 3a 20 3c 73 ;rport..From: <s
   69 70 3a 30 33 32 32 32 39 32 xx xx xx xx xx xx ip:032229xxxxxx@
   74 65 6c 2e 74 2d 6f 6e 6c 69 6e 65 2e 64 65 3e tel.t-online.de>
   3b 74 61 67 3d 61 73 31 39 62 62 63 38 33 36 0d ;tag=as19bbc836.
   0a 54 6f 3a 20 3c 73 69 70 3a 30 33 32 32 32 39 .To: <sip:032229
   xx xx xx xx xx xx 40 74 65 6c 2e 74 2d 6f 6e 6c xxxxxx@tel.t-onl
   69 6e 65 2e 64 65 3e 0d 0a 43 61 6c 6c 2d 49 44 ine.de>..Call-ID
   3a 20 37 66 63 36 63 62 66 32 34 30 34 35 62 62 : 7fc6cbf24045bb
   33 37 30 31 36 66 63 66 30 38 33 65 64 39 33 66 37016fcf083ed93f
   62 32 40 31 39 32 2e 31 36 38 2e 31 2e 32 30 30 b2@192.168.1.200
   0d 0a 43 53 65 71 3a 20 31 30 36 20 52 45 47 49 ..CSeq: 106 REGI
   53 54 45 52 0d 0a 55 73 65 72 2d 41 67 65 6e 74 STER..User-Agent
   3a 20 41 73 74 65 72 69 73 6b 20 50 42 58 0d 0a : Asterisk PBX..
   4d 61 78 2d 46 6f 72 77 61 72 64 73 3a 20 37 30 Max-Forwards: 70
   0d 0a 45 78 70 69 72 65 73 3a 20 33 36 30 30 0d ..Expires: 3600.
   0a 43 6f 6e 74 61 63 74 3a 20 3c 73 69 70 3a 30 .Contact: <sip:0
   33 32 32 32 39 32 39 33 39 31 33 40 38 34 2e 31 32229293913@84.1
   36 35 2e xx xx 2e xx xx xx 3e 0d 0a 45 76 65 6e 65.xx.xxx>..Even
   74 3a 20 72 65 67 69 73 74 72 61 74 69 6f 6e 0d t: registration.
   0a 43 6f 6e 74 65 6e 74 2d 4c 65 6e 67 74 68 3a .Content-Length:
   20 30 0d 0a 0d 0a                                0....

---end of BUFFER DUMP---

config:

if_inbound  = hme0
if_outbound = pppoe0
hosts_allow_reg = 192.168.1.0/24
hosts_allow_sip = 192.168.1.0/24
sip_listen_port = 5060
daemonize = 0
silence_log = 0
log_calls = 1
user = nobody
registration_file = /var/run/siproxd/siproxd_registrations
pid_file = /var/run/siproxd/siproxd.pid
rtp_proxy_enable = 1
rtp_port_low  = 10000
rtp_port_high = 20000
rtp_timeout = 300
default_expires = 600
debug_level = 0xffffffff
debug_port = 0

[Siproxd-users] Problems with incoming calls & Asterisk

[Julius S. <ju...@zg...>]

I've got the following set-up going now:
Asterisk <-> NAT router <-> internet
siproxd is running on the router. I've configured Asterisk like this in 
sip.conf:

[general]
outboundproxy=192.168.0.1
outboundproxyport=5060

register => julius:pass@<router (siproxd) ip>

for all the users I need

Outgoing calls all work perfectly with this set-up. Also most incoming 
calls are working fine, but not always though. Looking at the logs, it 
seems the party trying to call me, sometimes is handed the internal IP 
address of my Asterisk box. This results in voice going only one way. 
Any idea what could be going wrong here?
If you want to test this, you can call me on 
sip:ju...@sc...

Thanks a lot in advance,
Julius

Re: [Siproxd-users] Default forward

[Julius S. <ju...@zg...>]

Sorry, for the late reply.
Must say that the "Currently" in your message is interesting ;)
You're thinking of adding such a feature? It would solve a problem I've 
got very nicely.
Other than that siproxd is working _very_ well for me!

Thanks a lot for all the effort,

Julius

Thomas Ries schreef:
> Hi,
> 
> Currently, such a feature is not present in siproxd.
> 
> Regards,
> /Thomas
> 

Re: [Siproxd-users] Default forward

[Thomas R. <tr...@gm...>]

Hi,

Currently, such a feature is not present in siproxd.

Regards,
/Thomas

On  3 Dec, Julius Schwartzenberg wrote:
> Hi,
> 
> I was wondering if it was possible to have siproxd forward calls for all 
> addresses that aren't used by a registrar client to a certain host 
> that's inside my LAN instead of rejecting those. Is it possible to do 
> that with siproxd?
> 
> Thanks in advance,
> Julius

-- 
GnuPG: pub  1024D/87BCDC94 2000-03-19 Thomas Ries <tr...@gm...>
VoIP:  sip:174...@pr... | sip:43...@fw...

Do you know about the dangers of DRM?
Find out at http://www.defectivebydesign.org/what_is_drm

[Siproxd-users] Default forward

[Julius S. <ju...@zg...>]

Hi,

I was wondering if it was possible to have siproxd forward calls for all 
addresses that aren't used by a registrar client to a certain host 
that's inside my LAN instead of rejecting those. Is it possible to do 
that with siproxd?

Thanks in advance,
Julius

[Siproxd-users] Ekiga Configuration

[Ivan R. <iv...@gm...>]

Hello this si my network diagram

192.168.1.200(ekiga) ----> Linksys Nat to ----> 10.6.193.33 ---->
10.6.250.21(ADSL, siproxd)

I can't connect my ekiga softphone to ekiga.net, the request arrive to the
10.6.250.21 host
but after that nothing happens this is the log.

15:07:06 INFO:log.c:177 Accepted DEBUG TCP connection [fd=8]
15:07:06 INFO:log.c:178 siproxd-0.5.11-3017 i386-pc-linux-gnu
15:07:09 register.c:290 register: costal@192.168.1.200 expires=3600 seconds
15:07:09 register.c:315 found entry for costal@192.168.1.200 <->
co...@ek... at slot=0, exp=3005
15:07:09 register.c:415 masquerading UA costal@192.168.1.200 local
costal@189.145.28.195
15:07:10 register.c:290 register: costal@192.168.1.200 expires=3600 seconds
15:07:10 register.c:315 found entry for costal@192.168.1.200 <->
co...@ek... at slot=0, exp=3629
15:07:10 register.c:415 masquerading UA costal@192.168.1.200 local
costal@189.145.28.195
15:07:11 register.c:290 register: costal@192.168.1.200 expires=3600 seconds
15:07:11 register.c:315 found entry for costal@192.168.1.200 <->
co...@ek... at slot=0, exp=3629
15:07:11 register.c:415 masquerading UA costal@192.168.1.200 local
costal@189.145.28.195
15:07:13 register.c:290 register: costal@192.168.1.200 expires=3600 seconds
15:07:13 register.c:315 found entry for costal@192.168.1.200 <->
co...@ek... at slot=0, exp=3628
15:07:13 register.c:415 masquerading UA costal@192.168.1.200 local
costal@189.145.28.195
15:11:33 register.c:489 auto-saving registration table

Please help me !

[Siproxd-users] Is it possible to support tcp protocol for siproxd

[Praveenkumar P. - T. , C. <Pra...@hc...>]

    Hi,
      Actually siproxd supports only udp protocol,
      It is possible to support tcp protocol for siproxd.
      If we can support tcp protocol, what are the comlexites will be =
there?
      Where we have to change the code?
      can u give me the High end design to support tcp protocol for =
siproxd.
   =20

      thanks,
      praveen.
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Re: [Siproxd-users] max. concurrent calls

[Thomas R. <tr...@gm...>]

Ja, ich spreche auch Deutsch. Ich leben in der Schweiz.

/Thomas

On 19 Aug, Sebastian Gabris wrote:
> Hi,
>=20
> thanks for your answer Thomas. I will take a closer look at your programm=
.
>=20
> By the way do you speak german?
>=20
> Best regards,
>=20
> Sebastian
>=20
>=20
>> -----Urspr=FCngliche Nachricht-----
>> Von: Siproxd-users <sip...@li...>
>> Gesendet: 19.08.06 01:14:47
>> An: Siproxd-users <sip...@li...>
>> Betreff: Re: [Siproxd-users] max. concurrent calls
>=20
>=20
>>=20
>> It Depends.
>>=20
>> First thing is, you need a sufficient large port range defined for RTP
>> streams. Each stream requires 2 ports. The default values of
>>   rtp_port_low  =3D 7070
>>   rtp_port_high =3D 7089
>> would limit to 10 simulatneous RTP media streams (=3D 10 audio-only phon=
e
>> calls, 5 audio+video calls, you get the picture).
>>=20
>> Second, in siproxd.h are some constant values that are used for static
>> array sizing:
>>=20
>> URLMAP_SIZE   (128)=09max. number of clients that can be masqueraded
>> RTPPROXY_SIZE (256)=09max number of RTP datastreams
>>=20
>>=20
>> Third, performance. Currently there is (to my knowledge) no experience
>> available.
>>=20
>> /Thomas
>>=20
>>=20
>> On 18 Aug, Sebastian Gabris wrote:
>> > Hi all,
>> >=20
>> > i'm new to siproxd and would like to know how much concurrent calls do=
es siproxd  support.
>> >=20
>> > Thanks in advance :-)
>> >=20
>> > Best regards,
>> >=20
>> > Sebastian Gabris
>> > _____________________________________________________________________
>> > Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen!
>> > http://smartsurfer.web.de/?mc=3D100071&distributionid=3D000000000071
>> >=20
>> >=20
>> > ----------------------------------------------------------------------=
---
>> > Using Tomcat but need to do more? Need to support web services, securi=
ty?
>> > Get stuff done quickly with pre-integrated technology to make your job=
 easier
>> > Download IBM WebSphere Application Server v.1.0.1 based on Apache Gero=
nimo
>> > http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&da=
t=3D121642
>> > _______________________________________________
>> > Siproxd-users mailing list
>> > Sip...@li...
>> > https://lists.sourceforge.net/lists/listinfo/siproxd-users
>>=20
>> --=20
>> GnuPG: pub  1024D/87BCDC94 2000-03-19 Thomas Ries <tr...@gm...>
>> -      Fingerprint =3D 13D1 19F5 77D0 4CEC 8D3F  A24E 09FC C18A 87BC DC9=
4
>> -      Key via pgp.openpkg.org / http://www.ries.ch.vu/87BCDC94.pub
>> VoIP:  sip:174...@pr... | sip:43...@fw...
>> <hr>
>> ------------------------------------------------------------------------=
-
>> Using Tomcat but need to do more? Need to support web services, security=
?
>> Get stuff done quickly with pre-integrated technology to make your job e=
asier
>> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geroni=
mo
>> http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&dat=
=3D121642
>> <hr>
>> _______________________________________________
>> Siproxd-users mailing list
>> Sip...@li...
>> https://lists.sourceforge.net/lists/listinfo/siproxd-users
>>=20
>=20
>=20
> ______________________________________________________________
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> Jetzt bei WEB.DE FreeMail: http://f.web.de/?mc=3D021193
>=20
>=20
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> Get stuff done quickly with pre-integrated technology to make your job ea=
sier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronim=
o
> http://sel.as-us.falkag.net/sel?cmd=3Dlnk&kid=3D120709&bid=3D263057&dat=
=3D121642
> _______________________________________________
> Siproxd-users mailing list
> Sip...@li...
> https://lists.sourceforge.net/lists/listinfo/siproxd-users

--=20
GnuPG: pub  1024D/87BCDC94 2000-03-19 Thomas Ries <tr...@gm...>
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-      Key via pgp.openpkg.org / http://www.ries.ch.vu/87BCDC94.pub
VoIP:  sip:174...@pr... | sip:43...@fw...

Re: [Siproxd-users] max. concurrent calls

[Sebastian G. <Seb...@we...>]

Hi,

thanks for your answer Thomas. I will take a closer look at your programm.=


By the way do you speak german=3F

Best regards,

Sebastian


> -----Urspr=FCngliche Nachricht-----
> Von: Siproxd-users <sip...@li...>
> Gesendet: 19.08.06 01:14:47
> An: Siproxd-users <sip...@li...>
> Betreff: Re: [Siproxd-users] max. concurrent calls


>=20
> It Depends.
>=20
> First thing is, you need a sufficient large port range defined for RTP
> streams. Each stream requires 2 ports. The default values of
>   rtp=5Fport=5Flow  =3D 7070
>   rtp=5Fport=5Fhigh =3D 7089
> would limit to 10 simulatneous RTP media streams (=3D 10 audio-only phone
> calls, 5 audio+video calls, you get the picture).
>=20
> Second, in siproxd.h are some constant values that are used for static
> array sizing:
>=20
> URLMAP=5FSIZE   (128)	max. number of clients that can be masqueraded
> RTPPROXY=5FSIZE (256)	max number of RTP datastreams
>=20
>=20
> Third, performance. Currently there is (to my knowledge) no experience
> available.
>=20
> /Thomas
>=20
>=20
> On 18 Aug, Sebastian Gabris wrote:
> > Hi all,
> >=20
> > i'm new to siproxd and would like to know how much concurrent calls do=
es siproxd  support.
> >=20
> > Thanks in advance :-)
> >=20
> > Best regards,
> >=20
> > Sebastian Gabris
> > =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
> > Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen!
> > http://smartsurfer.web.de/=3Fmc=3D100071&distributionid=3D000000000071
> >=20
> >=20
> > ----------------------------------------------------------------------=
---
> > Using Tomcat but need to do more=3F Need to support web services, securi=
ty=3F
> > Get stuff done quickly with pre-integrated technology to make your job=
 easier
> > Download IBM WebSphere Application Server v.1.0.1 based on Apache Gero=
nimo
> > http://sel.as-us.falkag.net/sel=3Fcmd=3Dlnk&kid=3D120709&bid=3D263057&dat=3D1216=
42
> > =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
> > Siproxd-users mailing list
> > Sip...@li...
> > https://lists.sourceforge.net/lists/listinfo/siproxd-users
>=20
> --=20
> GnuPG: pub  1024D/87BCDC94 2000-03-19 Thomas Ries <tr...@gm...>
> -      Fingerprint =3D 13D1 19F5 77D0 4CEC 8D3F  A24E 09FC C18A 87BC DC94
> -      Key via pgp.openpkg.org / http://www.ries.ch.vu/87BCDC94.pub
> VoIP:  sip:174...@pr... | sip:43...@fw...
> <hr>
> ------------------------------------------------------------------------=
-
> Using Tomcat but need to do more=3F Need to support web services, security=
=3F
> Get stuff done quickly with pre-integrated technology to make your job e=
asier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geroni=
mo
> http://sel.as-us.falkag.net/sel=3Fcmd=3Dlnk&kid=3D120709&bid=3D263057&dat=3D121642=

> <hr>
> =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
> Siproxd-users mailing list
> Sip...@li...
> https://lists.sourceforge.net/lists/listinfo/siproxd-users
>=20


=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
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Re: [Siproxd-users] max. concurrent calls

[Thomas R. <tr...@gm...>]

It Depends.

First thing is, you need a sufficient large port range defined for RTP
streams. Each stream requires 2 ports. The default values of
  rtp_port_low  = 7070
  rtp_port_high = 7089
would limit to 10 simulatneous RTP media streams (= 10 audio-only phone
calls, 5 audio+video calls, you get the picture).

Second, in siproxd.h are some constant values that are used for static
array sizing:

URLMAP_SIZE   (128)	max. number of clients that can be masqueraded
RTPPROXY_SIZE (256)	max number of RTP datastreams


Third, performance. Currently there is (to my knowledge) no experience
available.

/Thomas


On 18 Aug, Sebastian Gabris wrote:
> Hi all,
> 
> i'm new to siproxd and would like to know how much concurrent calls does siproxd  support.
> 
> Thanks in advance :-)
> 
> Best regards,
> 
> Sebastian Gabris
> _____________________________________________________________________
> Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen!
> http://smartsurfer.web.de/?mc=100071&distributionid=000000000071
> 
> 
> -------------------------------------------------------------------------
> Using Tomcat but need to do more? Need to support web services, security?
> Get stuff done quickly with pre-integrated technology to make your job easier
> Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
> http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
> _______________________________________________
> Siproxd-users mailing list
> Sip...@li...
> https://lists.sourceforge.net/lists/listinfo/siproxd-users

-- 
GnuPG: pub  1024D/87BCDC94 2000-03-19 Thomas Ries <tr...@gm...>
-      Fingerprint = 13D1 19F5 77D0 4CEC 8D3F  A24E 09FC C18A 87BC DC94
-      Key via pgp.openpkg.org / http://www.ries.ch.vu/87BCDC94.pub
VoIP:  sip:174...@pr... | sip:43...@fw...

[Siproxd-users] max. concurrent calls

[Sebastian G. <Seb...@we...>]

Hi all,

i'm new to siproxd and would like to know how much concurrent calls does siproxd  support.

Thanks in advance :-)

Best regards,

Sebastian Gabris
_____________________________________________________________________
Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen!
http://smartsurfer.web.de/?mc=100071&distributionid=000000000071

[Siproxd-users] SIP client && siproxd on one laptop

[Matthias A. <gu...@Si...>]

Hello,

I want to use a SIP client (Ekiga) on my laptop and for the problems
which are causing the NAT issues someone recommended me to use siproxd.

The network scenario is like this:


 private IP address range             :          Internet
 192.168.2.x                          :          (public IP address range)
                                      :
                                      :         foo.bar.org
 +-------------+               +--------------+
 |             | .3         .1 | W-LAN WAN    | publicIP 82.135.67.x
 | laptop      |---------------| router       |------------>>
 |             |               |              |
 +-------------+               +--------------+
                 iwi0      eth0       :        ppp

and on 'laptop' with (private) IP 192.168.2.3 I want to use
the SIP client.

I've installed and configured the siproxd and it is up and running.
The comment in the config file of siproxd says what to do if the
siproxd is not running on the NAT gateway itself:

# The interface names of INBOUND and OUTBOUND interface.
#
#    If siproxd is not running on the host doing the masquerading
#    but on a host within the private network segment, "in front" of
#    the masquerading router: define if_inbound and if_outbound to
#    point to the same interface (the inbound interface). In *addition*
#    define 'host_outbound' to hold your external (public) IP address
#    or a hostname that resolves to that address (use a dyndns address for
#    example).
#
if_inbound  = iwi0
if_outbound = iwi0
# uncomment the following line ONLY IF YOU KNOW WHAT YOU ARE DOING!
# READ THE FAQ FIRST!
host_outbound = 82.135.1.6

The addr 82.135.1.6 is what my W-LAN Internet router got from the
provider. When I do a SIP call with Ekiga to sip:50...@ek... the
call is handled by siproxd but siproxd does not send it out to 
Ekiga.net:

# tcpdump -X -n -i lo0 port 5060
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
listening on lo0, link-type NULL (BSD loopback), capture size 96 bytes
10:41:19.454886 IP 192.168.2.3.5073 > 127.0.0.1.5060: SIP, length: 1022
        0x0000:  4500 041a 6c7d 0000 4011 c8a9 c0a8 0203  E...l}..@.......
        0x0010:  7f00 0001 13d1 13c4 0406 6ef8 494e 5649  ..........n.INVI
        0x0020:  5445 2073 6970 3a35 3030 4065 6b69 6761  TE.sip:500@ekiga
        0x0030:  2e6e 6574 2053 4950 2f32 2e30 0d0a 526f  .net.SIP/2.0..Ro
        0x0040:  7574 653a 203c 7369 703a 3132 372e 302e  ute:.<sip:127.0.
        0x0050:  302e 313a 3530 3630 3b6c 723e            0.1:5060;lr>
10:41:19.503308 IP 192.168.2.3.5060 > 192.168.2.3.5073: SIP, length: 340
        0x0000:  4500 0170 6c7f 0000 4011 87a7 c0a8 0203  E..pl...@.......
        0x0010:  c0a8 0203 13c4 13d1 015c 1fed 5349 502f  .........\..SIP/
        0x0020:  322e 3020 3430 3820 5265 7175 6573 7420  2.0.408.Request.
        0x0030:  5469 6d65 6f75 740d 0a56 6961 3a20 5349  Timeout..Via:.SI
        0x0040:  502f 322e 302f 5544 5020 3139 322e 3136  P/2.0/UDP.192.16
        0x0050:  382e 322e 333a 3530 3733 3b62            8.2.3:5073;b
10:41:19.504968 IP 192.168.2.3.5073 > 127.0.0.1.5060: SIP, length: 448
        0x0000:  4500 01dc 6c80 0000 4011 cae4 c0a8 0203  E...l...@.......
        0x0010:  7f00 0001 13d1 13c4 01c8 a53f 4143 4b20  ...........?ACK.
        0x0020:  7369 703a 3530 3040 656b 6967 612e 6e65  sip:50...@ek...
        0x0030:  7420 5349 502f 322e 300d 0a52 6f75 7465  t.SIP/2.0..Route
        0x0040:  3a20 3c73 6970 3a31 3237 2e30 2e30 2e31  :.<sip:127.0.0.1
        0x0050:  3a35 3036 303b 6c72 3e0d 0a43            :5060;lr>..C
10:41:19.505387 IP 192.168.2.3.5060 > 192.168.2.3.5073: SIP, length: 337
        0x0000:  4500 016d 6c81 0000 4011 87a8 c0a8 0203  E..ml...@.......
        0x0010:  c0a8 0203 13c4 13d1 0159 6ca7 5349 502f  .........Yl.SIP/
        0x0020:  322e 3020 3430 3820 5265 7175 6573 7420  2.0.408.Request.
        0x0030:  5469 6d65 6f75 740d 0a56 6961 3a20 5349  Timeout..Via:.SI
        0x0040:  502f 322e 302f 5544 5020 3139 322e 3136  P/2.0/UDP.192.16
        0x0050:  382e 322e 333a 3530 3733 3b62            8.2.3:5073;b

I've also enabled the debugging in the the siproxd and yes
they know that it should be send out, but it does not:

# tail -f /var/log/debug.log
Jun 25 10:41:19 rebelion siproxd: route_processing.c:134 route_preprocess: checking topmost Route header
Jun 25 10:41:19 rebelion siproxd: proxy.c:344 request [INVITE] from/to unregistered UA (RQ: gur...@ek... -> 50...@ek...)
Jun 25 10:41:19 rebelion siproxd: sock.c:164 send UDP packet to 192.168.2.3: 5073
Jun 25 10:41:19 rebelion siproxd: sock.c:125 received UDP packet from 192.168.2.3, count=448
Jun 25 10:41:19 rebelion siproxd: siproxd.c:362 checking Max-Forwards (=70)
Jun 25 10:41:19 rebelion siproxd: siproxd.c:408 received SIP type REQ:ACK
Jun 25 10:41:19 rebelion siproxd: proxy.c:88 proxy_request
Jun 25 10:41:19 rebelion siproxd: route_processing.c:134 route_preprocess: checking topmost Route header
Jun 25 10:41:19 rebelion siproxd: proxy.c:344 request [ACK] from/to unregistered UA (RQ: gur...@ek... -> 50...@ek...)
Jun 25 10:41:19 rebelion siproxd: sock.c:164 send UDP packet to 192.168.2.3: 5073

What I'm missing here?

	matthias

-- 
Matthias Apitz
Manager Technical Support - OCLC PICA GmbH
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t +49-89-61308 351 - f +49-89-61308 399 - m +49-170-4527211
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[Siproxd-users] poor sound quality on the echo test

[Joshua N P. <jpr...@po...>]

Hi!

1. I'm running siproxd 1:0.5.11-1 (debian) on my NAT/Firewall.

2. When I try kphone 1:4.0.5-2, everything seems to work but I get very
poor quality sound on the FWD echo test.  Sound quality on a one-way
test is fine.  I tried linphone 1.3.3-1 and the sound quality was worse.
I could hardly hear anything on the echo test.

3. My ping time to fwd sucks:

joshua@emit:~$ ping fwd.pulver.com
PING fwd.pulver.com (69.90.155.70): 56 data bytes
64 bytes from 69.90.155.70: icmp_seq=3D0 ttl=3D44 time=3D371.5 ms
64 bytes from 69.90.155.70: icmp_seq=3D1 ttl=3D44 time=3D368.1 ms

4. I've tried with asterisk and a Grandstream phone instead of siproxd.
It works fine, echo test and all.  However, I don't want to run asterisk
if I can run siproxd instead.  Siproxd uses about 10% of the memory that
asterisk wants.  I don't need the extra features.

5. I haven't tried the Grandstream + siproxd because my phone is out for
repair.  Similarly, I couldn't get kphone to talk to asterisk for some
reason.  If nobody can suggest a solution then I'll test more
combinations.

I suspect that the only problem is that siproxd is not coping
with high latency.  Is there any config option to get better behavior
with high latency?

--=20
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