Thanks, Michael. By the way, do you know how to convert a sip_t pointer into a
msg_t pointer? My callback function gets a sip_t pointer.
Regards,
Jerry
From: Michael Jerris [mailto:m...@jerris.com]
Sent: Thursday, May 08, 2014 11:07 AM
To: Jerry Richards
Cc: sofia-sip-devel
Hello,
I'm trying to handle an in-dialog NOTIFY Event hold/talk message (i.e. to put a
call on-hold and take off-hold) from a SIP server. The NOTIFY has no
subscription and sofia-sip reports the nua_i_notify event with the same
nua_handle as the INVITE.
The problem is when I invoke nua_respon
Hello,
I have sofia-sip setup with one NUA agent (i.e. one call to nua_create()). Is
there a way to send a REGISTER message to a proxy different than the binding
established via nua_create()? The reason is, in a failover condition, I want
to find an easy way to periodically check if it can sw
don't know is how to obtain the actual TCP/IP source address within
sofia-sip (I only know how to access the content of the message, not the TCP/IP
header information).
Thanks,
Jerry
From: Michael Jerris [mailto:m...@jerris.com]
Sent: Wednesday, March 27, 2013 9:30 AM
To: Jerry Richar
Thanks, Mike. Do you know what those few lines would be? How can sofia-sip
know the INVITE is not spoofed from another entity?
Jerry
From: Michael Jerris [mailto:m...@jerris.com]
Sent: Tuesday, March 26, 2013 3:32 PM
To: Jerry Richards
Cc: 'sofia-sip-devel@lists.sourceforge.net'
S
Hello,
As a phone endpoint (i.e. user agent), does sofia-sip have an option tag to
reject any SIP requests/messages not coming from the SIP proxy to which it is
registered?
Best Regards,
Jerry
--
Own the Future-Intel®
The SIP OPTIONS CSeq is not incrementing when I include NUTAG_KEEPALIVE(25000)
with the nua_register() command. Is there a parameter to increment the CSeq
for each OPTION message transmitted?
Thanks,
Jerry
--
Monitor y
I have a Freeswitch issue, where I have softphones connected to two separate
subnets (one on eth0 and the other on eth1). When I send an IM (i.e. SIP
MESSAGE) from the eth1 phone to the eth0 phone, a wireshark trace shows the
source IP address of the eth1 server address, rather than the eth0 se
nyone know how to do this?
Thanks,
Jerry
From: Jerry Richards
Sent: Monday, February 20, 2012 11:04 AM
To: 'sofia-sip-devel@lists.sourceforge.net'
Subject: RE: Port 1024 Open?
Okay, I found that TPTAG_SERVER is enabled by default. This was causing port
1024 to be opened when when usi
Okay, I found that TPTAG_SERVER is enabled by default. This was causing port
1024 to be opened when when using TLS. Why is this enabled by default? I'm
using sofia-sip for a phone application so this definitely needs to be disabled.
Thanks,
Jerry
From: Jerry Richards
Sent: Tuesday, Feb
Is it possible that sofia-sip is opening port 1024? We did a port scan on our
phone and port 1024 is open, but I can't find what is opening it. It's either
sofia-sip or something else?
Thanks,
Jerry
--
Keep Your Devel
Actually, I think this was a glitch in a log capture, so this might be a
non-issue.
From: Jerry Richards
Sent: Wednesday, September 28, 2011 3:57 PM
To: 'sofia-sip-devel@lists.sourceforge.net'
Subject: Sofia-sip Firing nua_r_subscribe Event Instead of nua_r_unsubscribe
Event
Has anyo
Has anyone noticed sofia-sip generating the nua_r_subscribe event instead of
nua_r_unsubscribe event, after invoking nua_unsubscribe()?
Thanks,
Jerry
--
All the data continuously generated in your IT infrastructure contai
cr_method ==
sip_method_bye));
The OLD line was causing a SIGNAL ( 6:SIGIOT/SIGABRT:IOT trap/Abort) for
pthread su_root_run in my phone.
Thanks,
Jerry
From: Jerry Richards
Sent: Wednesday, June 22, 2011 9:07 AM
To: 'sofia-sip-devel@lists.sourceforge.net'
Subject: Segmentation Fault
If you recommend a different fix, please let me know.
Thanks,
Jerry
From: Jerry Richards
Sent: Friday, June 17, 2011 9:33 AM
To: 'Pekka Pessi'
Subject: FW: tls_connect() Invoked On Every Outbound INVITE
Hi Pekka,
The sofia-sip mailing list did not like my message below because it wa
Hello,
Anyone know why sofia-sip would re-establish a TLS connection for every
outbound INVITE? That's what I see happening. Logs are shown below, after
calling nua_invite() while a TLS connection was already establised (Note:
you'll probably see some extra logs I added to help debug this).
We pulled version 1.12.11 and did some load testing. I have an automated test
that makes 2 phones complete calls to 1 phone iteratively. We noticed that the
callers are losing a lot of memory over time.
Using the previous sofia-sip version (1.12.9) each caller could make about
35,000 calls be
Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] Setting sofia-sip Source Port
Hi Jerry,
2011/4/4 Jerry Richards :
> I temporarily got it working by patching the tport_tls_connect() function to
> set susa.su_port to 5061 (instead of setting it to 0). I do
not a random port).
Thanks,
Jerry
-Original Message-
From: Pekka Pessi [mailto:ppe...@gmail.com]
Sent: Monday, April 04, 2011 8:35 AM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] Setting sofia-sip Source Port
Hi,
2011/3/31 Jerry Richards :
&
Hello,
I am setting my contact address to specify port 5061 (for TLS), but a wireshark
trace shows the message being sent with a source port of 1025. This is
confusing the Redcom server. How can I make sofia-sip send the message with
the same port as is in the contact header?
Thanks,
Jerry
-
FYI, I found a typo (i.e. bug) in sofia-sip version 1.12.11. Line #121 of
soa_tag.h should be:
SOFIAPUBVAR tag_typedef_t soatag_local_sdp_str_ref;
Jerry
--
Create and publish websites with WebMatrix
Use the most popula
Pekka,
I pulled the version using darcs. If you're now using git, what would be the
command line to fetch the latest source (using git)?
Thanks,
Jerry
-Original Message-
From: Pekka Pessi [mailto:ppe...@gmail.com]
Sent: Thursday, March 24, 2011 7:50 AM
To: Jerry Richards
Cc:
March 24, 2011 6:09 AM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] Good TLS Hellos But No SIP REGISTER Sent
Hi Jerry,
2011/3/22 Jerry Richards :
> I am seeing an issue with TLS where the REGISTER message is not
> getting sent. The TLS Hello/
I noticed the latest trunk still has the following line in the configure.ac
file:
AC_INIT([sofia-sip], [1.12.10devel])
Shouldn't this line show 1.12.11?
Thanks,
Jerry
--
Enable your software for Intel(R) Active Manageme
Hello,
I am seeing an issue with TLS where the REGISTER message is not getting sent.
The TLS Hello/Certificate exchange seems to always happens correctly (which
just precedes the REGISTER request). The scenario is that it is previously
registered and then later lose registration because of th
Hello,
I have a server that is closing the TLS connection if it receives no messages
after some time. My phone retries registering every 40 seconds, but I notice
it does not re-connect the TLS every 40 seconds (it just assumes it's already
connected).
How do I tell sofia-sip to go through th
.
Thanks,
Jerry
From: Jerry Richards
Sent: Wednesday, February 02, 2011 12:59 PM
To: sofia-sip-devel@lists.sourceforge.net
Subject: Sofia Getting 900 Internal Error When
SIPTAG_TO_STR("sip:7348902;rfrid=eaff00221c6035b5@200.21.3.10")
Hello All,
Do you know why the sofia stack is reje
Hello All,
Do you know why the sofia stack is rejecting this SIPTAG_TO_STR() when calling
nua_invite()? If I remove the "rfid=x..." portion it gets no error. This
is an INVITE sent in response to a received REFER method.
SIPTAG_TO_STR("sip:7348902;rfrid=eaff00221c6035b5@200.21.3.10")
Hello,
I am cross-compiling sofia-sip and I'm wondering if anyone has gotten
sofia-sip to request type addresses in its DNS query? Right now I only
see it do a type A request and if that fails, it gives up. I would like it
to attempt type A and if that fails, attempt type .
I can see t
Hello,
SIP Contact Headers should always be the URL of the device sending the
message (i.e. the address at which the sender can be contacted/replied-to).
So I would always expect the Contact Header in the 200 OK to be the URL of
the sender (not the URL of the receiver). This is why I don't unders
Contact
URL to the received Contact URL which is always different (because the sent
URL is the phone's URL and the received URL is the server's URL), thus the
code that sets the refresh time to the received Expires value is never
getting invoked.
Jerry
-Original Message-----
F
Well, I went ahead and added the su_free() call and it appears to have
significantly improved my memory leak situation. However, the phone still
loses ~122 bytes per call (i.e. allocated but not freed).
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jer
Hello,
I am using version 1.12.9 and noticed the sofia-sip stack setting the
register refresh timer to the wrong value. That is, the 200 OK includes
"Expires: 60", but sofia-sip logs this:
"nua(): refresh register after 1905 seconds (in [900..2700])".
Here is the 200 OK content:
Status-Line: S
Hello,
Does sofia-sip support Incoming Call Screening per the following link?
http://www.tech-invite.com/Ti-sip-service-14.html#fig1
Best Regards,
Jerry
--
___
Sofia-sip-deve
H... After a bit more reading, I'm thinking in my example code snippet
(below), I need to use the following call:
su_free(appl->home, replaces_hdr);
True?
--
___
Sofia-si
Hello All,
If I call nua_handle_make_replaces(), must I free the return pointer? For
example, here is a code snippet:
sip_replaces_t *replaces_hdr = nua_handle_make_replaces(nh, appl->home,
false); ...
free(replaces_hdr); /* << Is this necessary? <<< */
The reason I ask is because I
Hello All,
I noticed if I use SIPTAG_REQUIRE_STR("timer"), then the outbound INVITE
does NOT include a "Session-Expires" header. However, if I use
SIPTAG_SUPPORTED_STR("timer"), then the outbound INVITE does include a
"Session-Expires" header. Also by the way, in both cases I set
NUTAG_SESSION_T
Oops, my bad. I was doing something else wrong. Sofia-sip is working
beautifully as always.
Best Regards,
Jerry
-Original Message-
From: Pekka Pessi [mailto:ppe...@gmail.com]
Sent: Thursday, June 25, 2009 11:09 AM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject
Hello All,
If I include an "Expires: 240" header line in my nua_invite() call, it
causes sofia-sip to generate the following error and the INVITE is not sent.
This happens if I use SIPTAG_HEADER_STR("Expires: 240") or
SIPTAG_EXPIRES_STR("240").
nua_r_invite event: 900 Internal error at
../../.
, 2009 5:10 AM
To: Jerry Richards
Subject: Re: [Sofia-sip-devel] Ordering of SDP Fields
2009/4/14 Jerry Richards :
> Do you know why SOATAG_ORDERED_USER(1) is not working for me? I tried
> adding it to the nua_create(), nua_invite(), and nua_register() calls,
> but any outbound INVITE
t;Require: 100rel" header.
Best Regards,
Jerry
-----Original Message-
From: Jerry Richards [mailto:jer...@tonecommander.com]
Sent: Friday, May 15, 2009 1:55 PM
To: 'Jerry Richards'; 'Pekka Pessi'
Cc: 'sofia-sip-devel@lists.sourceforge.net'
Subject: RE
Hello All,
I noticed if I attempt to register via TLS transport and the server does not
reply, then after 30 seconds the TLS connection is shutdown. Does anyone
know why? Is this expected upon a REGISTER timeout?
Best Regards,
Jerry
out the entire if-statement below, and now I see the 180 Ringing
get transmitted with the "Require: 100rel" header.
Are there any side-effects/pitfalls to doing this?
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jer...@tonecommander.com]
Pekka,
I commented out the entire if-statement below, and now I see the 180 Ringing
get transmitted with the "Require: 100rel" header.
Are there any side-effects/pitfalls to doing this?
Best Regards,
Jerry
-Original Message-----
From: Jerry Richards [mailto:jer...@tonecom
how to fix this?
Best Regards,
Jerry
-Original Message-
From: Pekka Pessi [mailto:ppe...@gmail.com]
Sent: Friday, May 15, 2009 9:29 AM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] "Require: 100rel" in 180 Ringing Message
2009/5/15 Jerry Richard
si [mailto:ppe...@gmail.com]
Sent: Friday, May 15, 2009 9:29 AM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] "Require: 100rel" in 180 Ringing Message
2009/5/15 Jerry Richards :
> If I invoke nua_respond(nh, 180, "Ringing",
&
Hello All,
I need to put a "Require: 100rel" header in a 180 Ringing message, but I am
having difficulties.
If I invoke nua_set_hparams(nh, SIPTAG_REQUIRE_STR("100rel")) prior to the
call to nua_respond(nh, 180, "Ringing"), then the 180 Ringing message does
not contain the "Require: 100rel" heade
[mailto:ppe...@gmail.com]
Sent: Monday, March 02, 2009 8:30 AM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] Ordering of SDP Fields
2009/2/27 Jerry Richards :
> When my application specifies the SDP in the SOATAG_USER_SDP(), I
> noticed sofia-sip som
Hello All,
I found a rare sofia bug when answering a "401 Authentication Required"
message if the realm is defined as an IPv6 address (e.g.
"scheme:\"[fe80::204:23ff:fea7:d60a]\":user:password"). The symptom was a
"904 No Matching Challenge" reply to the nua_authenticate().
I changed the followi
Hello,
You only need to call nua_authenticate() if the event status is 401 or 407.
Not sure if there are other reasons for the 904 in this case.
Jerry
--
This SF.net email is sponsored by:
High Quality Requirements in a
Jerry
-Original Message-
From: Jerry Richards [mailto:jer...@tonecommander.com]
Sent: Friday, March 13, 2009 8:03 AM
To: 'sofia-sip-devel@lists.sourceforge.net'
Cc: 'pekka.pe...@nokia.com'; 'mikhail.zabal...@nokia.com'
Subject: RE: 904 No Matching Challenge
Hello All,
Ok
Message-
>From: ext Jerry Richards [mailto:jer...@tonecommander.com]
>Sent: Thursday, March 12, 2009 10:57 PM
>To: Zabaluev Mikhail (Nokia-D/Helsinki)
>Cc: sofia-sip-devel@lists.sourceforge.net
>Subject: RE: 904 No Matching Challenge
>
>Could you send me a patch (or poi
Hi Mikhail,
Could you send me a patch (or point out the code) for what you described
below?
"When I removed the line that invalidated the cached challenge upon
authentication failure, ..."
Thanks
Jerry
--
Apps built w
Hi Martin,
Could you give me a patch for what you described below?
"When I removed the line that invalidated the cached challenge upon
authentication failure, ..."
Thanks,
Jerry
--
Apps built with the Adobe(R) Flex(R)
Hello All,
If my MD5 Authentication username is incorrect, I noticed my sofia-sip phone
will get a "904 No Matching Challenge" event in response to my
nua_authenticate() call. This only happens AFTER the first
nua_authenticate() call (the first one works correctly).
Has anyone encountered a simi
transport only.
2) In tport_base_connect(), just prior to the bind()
call, I set the port number to what is specified
by the NUA (not 0).
Now my application uses the specific source port for UDP, TCP, and TLS.
Jerry
-Original Message-
From: Jerry Richards [mailto:jer
or TLS
connection.
Best Regards,
Jerry
-Original Message-
From: Pekka Pessi [mailto:ppe...@gmail.com]
Sent: Thursday, February 26, 2009 2:36 PM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] TCP Source Port 3072?
2009/2/26 Jerry Richards :
Hello All,
When my application specifies the SDP in the SOATAG_USER_SDP(), I noticed
sofia-sip sometimes reorders the lines of the SDP in the message that is
actually transmitted. Do you know why this is? Does the order matter?
Best Regards,
Jerry
-
ginal Message-
From: Pekka Pessi [mailto:ppe...@gmail.com]
Sent: Thursday, February 26, 2009 12:15 PM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] TCP Source Port 3072?
2009/2/26 Jerry Richards :
> Hello All,
>
> When I configure the sofia trans
Pekka Pessi [mailto:ppe...@gmail.com]
Sent: Thursday, February 26, 2009 12:15 PM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] TCP Source Port 3072?
2009/2/26 Jerry Richards :
> Hello All,
>
> When I configure the sofia transport=udp, then I see S
Hello All,
When I configure the sofia transport=udp, then I see SIP UDP packets sent
with source port 5060 and destination port 5060. When the only change I
make is transport=tcp, then I see SIP TCP packets sent with source port 3072
and destination port 5060.
Has anyone seen this? Maybe I'm mi
Hello All,
Another issue related to nua_prack(): The RAck and RSeq headers are missing
in the outbound PRACK message.
Do you know why?
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jer...@tonecommander.com]
Sent: Wednesday, February 25, 2009 10:34 AM
To
Hello All,
When I invoke nua_prack() with the SOATAG_USER_SDP(), sofia-sip is not
including the SDP body in the outbound PRACK message. Here is the scenario:
NUA Server
| INVITE/SDP|
|>|
| 100 Trying|
|<|
| 183 Ringing/SDP |
|<--
Hello All,
Does anyone know of a reason why I cannot send messages from my phone having
an IPv4 address to a SIP server having an IPv6 address? We will all be
living in a world of dual IPv4/6 networks for quite a while, so I think this
should work. The function tport_vsend() logs the following e
ould be added to pass in the name of the interface or
index of the interface (unless there already is one)?
Best Regards,
Jerry
-Original Message-
From: Pekka Pessi [mailto:ppe...@gmail.com]
Sent: Wednesday, February 11, 2009 12:42 PM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforg
Hello All,
Anyone know why sofia-sip's tport UDP bind() invocation is happy if I use
the phone's IPv6 Global address, but gets an error (Invalid Argument) when I
use the phone's IPv6 Link address?
Best Regards,
Jerry
--
Hello All,
Could someone tell me generally what I need to change to send SIP messages
over an IPv6 network. Is it simply a matter of substituting the IPv6
address where ever I normally put the IPv4 address? Thus far, I have only
had my phone connected to an IPv4 network, but now would like to co
Hello,
I would vote for Subversion, mostly because we use that here where I work.
We've been using Subversion for about 3 years now and it has been reliable,
once you understand how it operates. Keep in mind we only have a handful of
engineers using it so it's easy to avoid conflicts, and therefo
-
From: Pekka Pessi [mailto:ppe...@gmail.com]
Sent: Tuesday, January 27, 2009 4:22 AM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] Port Number in Contact Header
2009/1/27 Jerry Richards :
> I noticed sofia-sip leaves out the port number i
Hello All,
I noticed sofia-sip leaves out the port number in the contact header. The
only way to include the port number in the Contact Header is to use the
SIPTAG_CONTACT_HEADER_STR() tag.
I'm just wondering if the sofia-sip default should be to include the port
number (e.g. ) instead of exclud
ssage-----
From: Jerry Richards [mailto:jer...@tonecommander.com]
Sent: Friday, January 23, 2009 2:46 PM
To: 'sofia-sip-devel@lists.sourceforge.net'
Subject: Call-Id Header Generation
Hello All,
Anyone know why sofia's SIP call-id is not of the form "loca...@host"? All
Hello All,
Anyone know why sofia's SIP call-id is not of the form "loca...@host"? All
I get is "localid". Unfortunately, I am testing against a new server and it
appears it wants "loca...@host". Is there a way to do this in sofia-sip?
Best Regards,
Jerry
-
By the way, regarding my Email below, I set the following tag during
initialization: NTATAG_UDP_MTU(2000). This should be sufficient for the
INVITE (which is of size ~1508).
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jer...@tonecommander.com]
Sent: Friday
Hello All,
Does anyone know why I would see "Fragmented IP protocol (proto=UDP 0x11,
Off=0)" frames in my Wireshark/Ethereal trace file? The message is an
outbound INVITE from my sofia phone, and Wireshark notes that it reassembled
the IP packet in the following frame (which is an INVITE). I not
Paulo,
We also discovered sofia-sip is not verifying TLS/SSL certificates. Could
you send me this patch? I don't see it up in darcs yet.
Best Regards,
Jerry
Message: 2
Date: Wed, 19 Nov 2008 18:46:37 -0200
From: "Paulo Pizarro" <[EMAIL PROTECTED]>
Subject: [Sofia-sip-devel] patch: TPTAG_TLS
Hello All,
Anyone know the answer to the question below? Does sofia-sip only use
"agent.pem" and "cafile.pem"? No other pem files are needed or specified?
I'm somewhat new to SSL/TLS.
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:[EMAIL
Hello All,
I'm setting NUTAG_SESSION_REFRESHER(nua_local_refresher) in a call to
nua_set_params() after the call to nua_create(). Outbound INVITEs are then
sent with the following header, as expected:
Session-Expires: 3600;refresher=uac
However, when "200 OK" is sent in reply to an INVITE, sofi
Hello All,
Is there an existing tag to include the "P-Early-Media: supported" header in
an outbound INVITE? If not, perhaps it should be added? Setting
NUTAG_EARLY_MEDIA(1) in the nua_invite() does not generate the P-Early-Media
header.
Best Regards,
Jerry
Hello All,
I am starting to test a TLS connection and I noticed there is the
TPTAG_CERTIFICATE() tag, which is described to be the "path to the public
key certificate directory". However, shouldn't there also be tags for the
key filename and CA filename? I noticed the tport_tls_init_master()
fun
Hello All,
Sorry if this is a repeat Email. I think I put the wrong To Header on my
last one.
I am in the process of upgrading from v1.12.7+ to v1.12.9+, and I noticed
something that no longer works the way it used to.
I have logic in my application to send an INVITE to a destination different
AM
To: [EMAIL PROTECTED]; sofia-sip-devel@lists.sourceforge.net
Subject: RE: [Sofia-sip-devel] sofia-sip/configure Script Error
>-Original Message-
>From: ext Jerry Richards [mailto:[EMAIL PROTECTED]
>Sent: Thursday, October 16, 2008 10:10 PM
>To: sofia-sip-devel@lists.sourcef
Hello,
I'm not sure if anyone noticed this also, but I found that line #26605
(v.1.12.9) of the sofia-sip/configure script gets an error, complaining that
it expects a unary operator. So I modified the line in the following way
and the error is corrected:
CHANGE: if test $HAVE_OPENSSL = 1 ; the
Hello,
Was a tag added for this? Please see Email below.
Best Regards,
Jerry
-Original Message-
From: Pekka Pessi [mailto:[EMAIL PROTECTED]
Sent: Monday, February 25, 2008 7:13 AM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] SUBSCRIBE
Hello,
Was a tag ever added for this? See last line of Email below.
Best Regards,
Jerry
-Original Message-
From: Pekka Pessi [mailto:[EMAIL PROTECTED]
Sent: Friday, February 01, 2008 5:51 AM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel
, July 10, 2008 11:03 AM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: Possible Memory leak in 1.12.9
Hi Jerry,
We have a sofia based UA application that has been running on 1.12.6 for
quite a while.
The application does call nua_handle_destroy(). It is in the debug print
the telephone-event line appears correctly in my SDP in the 200_OK.
Note: I also include the "a=rtpmap:101 telephone-event/8000" line in my
SOATAG_USER_SDP_STR() tag.
Best Regards,
Jerry
-Original Message-
From: Timo Bruhn [mailto:[EMAIL PROTECTED]
Sent: Monday, October 06, 200
Bruhn [mailto:[EMAIL PROTECTED]
Sent: Sunday, October 05, 2008 10:28 PM
To: Jerry Richards; sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] SIP Compact Headers
Hi,
you can use
NTATAG_SIPFLAGS(MSG_FLG_COMPACT)
in nua_create().
Unfortunately this way all messages are ge
Hello All,
I am using base version 1.12.7. Upon an inbound INVITE, I call
nua_respond() with the SOATAG_USER_SDP_STR() tag as follows:
o=TC 0 1 IN IP4 192.168.72.161
s=session
c=IN IP4 192.168.72.161
m=audio 1760 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
Ho
Hello All,
I think sofia-sip is capable of parsing inbound SIP messages with Compact
Headers, but I am not sure if I can configure it to generate SIP Compact
Headers.
Is there a tag to do this?
Best Regards,
Jerry
-
This S
Hello All,
Is there any reason why STUN is not mentioned in the Sofia-SIP features
table page, even though it is supported? Here's the link:
http://sofia-sip.sourceforge.net/refdocs/sofia_sip_conformance.html
Best Regards,
Jerry
Hello All,
When using SIP-URI formats, is the "logical recipient" in the Request-URI
and To-Header URI, always the same for REGISTER, and INVITE methods?
By logical recipient, I mean, the domain portion of the following:
"sip:[EMAIL PROTECTED]".
Best Regards,
Jerry
Hello All,
Okay, I did not hear any response to this issue, so I'll go ahead and
implement a refresh timer in my NUA application in the case that the server
indicates "refresher=uac" in the Session-Expires header (see below).
Best Regards,
Jerry
-Original Message-----
From:
Hello All,
Anyone know the answer to this issue below? I haven't heard back from
anyone.
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:[EMAIL PROTECTED]
Sent: Monday, August 25, 2008 3:19 PM
To: 'sofia-sip-devel@lists.sourceforge.net'
Subject:
Hello All,
It appears my sofia-sip NUA (based on v1.12.7) is not refreshing a session
IAW RFC 4028. Here is the scenario:
Sec Event
- ---
102: NUA sends INVITE with "Supported: timer", "Session-Expires: 3600"
111: UAS sends 200 OK with "Supported: timer" only
111: UAS sends re-I
Best Regards,
Jerry
-Original Message-
From: Pekka Pessi [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 14, 2008 4:19 AM
To: Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] General SIP Server Question
2008/8/13 Jerry Richards <[EMAIL PROTECTE
Hello All,
General question regarding a SIP phone configuration where the phone
communicates with a single SIP server. Would you agree such a phone need
only configure the following two SIP server entities?
1) SIP Registrar (either IP-address or domain-name) to be used inside the
SIP messages fo
rom: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of ext
>Jerry Richards
>Sent: Wednesday, July 23, 2008 1:42 AM
>To: sofia-sip-devel@lists.sourceforge.net
>Subject: [Sofia-sip-devel] TEL URI
>
>I am trying to send an INVITE using a tel URI instead of a SIP URI.
Hello All,
I am trying to send an INVITE using a tel URI instead of a SIP URI.
According to the following page, sofia-sip should support this:
http://sofia-sip.sourceforge.net/refdocs/sofia_sip_conformance.html
However, the following API calls result in a 900 Internal NUA Error at
../../../libso
Hello All,
Question on an issue I am facing. I have a case where I will reject a
re-INVITE under certain circumstances. The scenario follows:
1) Receive: re-INVITE
2) Callback: nua_i_invite event (100 Trying)
3) Send: 403 Forbidden (nua_respond)
4) Callback: nua_i_state event nua_callstate_
Hello All,
Is sofia-sip compliant with IMS? If so, what changes do I need to make in
my NUA application to make it compliant with IMS?
Best Regards,
Jerry
-
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