Hi Pete,
I have read through all your posts on the mailing list and I would
> like to implement the same 1 listener , multiple worker model that you
> are mentioning here.
>
> I have question about how to use the SU api to sprawn thread for each
> incoming INVITE. Can you please share with me the
Hi Aleksander,
I have read through all your posts on the mailing list and I would
like to implement the same 1 listener , multiple worker model that you
are mentioning here.
I have question about how to use the SU api to sprawn thread for each
incoming INVITE. Can you please share with me the st
Just to follow up on this. I modified the code to run in stateful mode and did
not encounter the problems I had in stateless. I also see that in stateful
mode the stack does generate a 100 Trying if the N1 timer expires, and that it
also handles an incoming CANCEL of a transaction by sending a
Hi, people -
I realized that many times when a server doesn't answer an OPTIONS request,
there's no retransmissions and
is reported this:
tport_udp_error: Connection refused (111) [icmp type=3 code=3]
reported by [10.20.40.201]:0
nta: OPTIONS (125368570): Connection refused (111) with
udp
Hi -
I am writing a fairly simply stateless redirect server. It is multithreaded,
where the main thread creates the root task and runs the thread loop, and
receives incoming SIP invites. If I simply respond to the INVITE with
nta_msg_treply in the main thread, everything works. However, if
>>There aren't successful INVITEs? What is it exactly what your app is doing?
Sorry, I was classifying any non-200 response as non-success. Mainly I am
returning 300 Multiple Choices responses.
On Jan 4, 2010, at 11:59 AM, Aleksander Morgado wrote:
> Hi Dave,
>
>
> 1) Sending the provision
Hi Dave,
> 1) Sending the provisional response definitely seems to mark the message
> for deletion, reclamation, whatever. I did a simple test of trying to send
> a final response from the main loop right after sending a 100 Trying and got
> the same "message could not be completed" message. So
Hi Aleksander,
I did a few tests, and this is what I learned:
1) Sending the provisional response definitely seems to mark the message for
deletion, reclamation, whatever. I did a simple test of trying to send a final
response from the main loop right after sending a 100 Trying and got the sa
Hi Dave,
Thanks Aleksander, I had actually seen your thread from some time ago and it
> was what inspired me to take this general approach. So thanks twice!
>
:-)
>
> I have figured out some of my problem, and also have a few questions about
> some of the suggestions below. First, after fiddl
Thanks Aleksander, I had actually seen your thread from some time ago and it
was what inspired me to take this general approach. So thanks twice!
I have figured out some of my problem, and also have a few questions about some
of the suggestions below. First, after fiddling with my code, I see
HI Dave,
> I am writing a fairly simply stateless redirect server. It is
> multithreaded, where the main thread creates the root task and runs the
> thread loop, and receives incoming SIP invites. If I simply respond to the
> INVITE with nta_msg_treply in the main thread, everything works. How
Hi -
I am writing a fairly simply stateless redirect server. It is multithreaded,
where the main thread creates the root task and runs the thread loop, and
receives incoming SIP invites. If I simply respond to the INVITE with
nta_msg_treply in the main thread, everything works. However, if
2009/9/9 Daniel Corbe :
> When NTA receives a request which doesn't match an existing transaction it
> passes a pointer to the callback function of type sip_t. Can this pointer be
> dereferenced through the lifetime of the transaction or does it go poof as
> soon as I exit the callback function
Hi Daniel,
> When NTA receives a request which doesn't match an existing transaction it
> passes a pointer to the callback function of type sip_t. Can this pointer
> be dereferenced through the lifetime of the transaction or does it go poof
> as soon as I exit the callback function?
>
>
My under
Hello list,
When NTA receives a request which doesn't match an existing transaction it
passes a pointer to the callback function of type sip_t. Can this pointer be
dereferenced through the lifetime of the transaction or does it go poof as soon
as I exit the callback function?
I ask because I'
Hi,
a bit late reply but here goes anyways.
On 30 June 2009, Luo Cheng wrote:
>Sorry for dump to much questions and emails about sofsip-cli here!
no problem, that's what the list is for! :)
>I was trying to solve the audio call problems with
>"priv_cb_pipeline_bus: Error: Could not get/set s
Hi,
Sorry for dump to much questions and emails about sofsip-cli here!
I was trying to solve the audio call problems with "priv_cb_pipeline_bus:
Error: Could not get/set settings from/on resource" error.
Basically callback func: -
static gboolean priv_cb_pipeline_bus (GstBus *bus, GstMessage *
Hi,
I have a problem with sofia-sip usage in Freeswitch. Freeswitch uses
sofia-sip and the problem I am having is related to running > 250 cps.
The test scenario is with sipp uac on one side sending calls to uas on the
other side of Freeswitch. Uas waits for 5 seconds and hands up.
This scenari
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
Hello,
i' ve tried to write something like this in my callback function:
void app_callback(nua_event_t event,
int status,
char const *phrase,
nua_t *nua,
nua_magic_t *magic,
nua_handle_t *nh,
2009/2/6 Andrew Rechenberg Lists :
> OK, I've added code to nta.c for the SRV records, however this line is
> tripping me up:
>
> if (strcmp(tpname, "*")) /* Looking for only one transport */
> break;
>
> I've dug through the code and can't find out how to set this tpname. Is
> there an NTATAG th
> >
> > Anywhere else I need to add code? Any suggestions on what I should
> add?
> > Can I add things like:
> >
> > { "tls-ms-i", "5061", "_sipinternaltls._tcp.", "SIPS+D2T" }
> >
> > Or should I just add another "tls" entry with the _sipinternaltls
> > portion?
>
> I think another "tls" entry i
> > I'm wondering how I should add this code to Sofia. I think I should
> be
> > adding it to nta.c in const sip_dnstports[] and possibly in
> > tports_sips[].
> >
> > Anywhere else I need to add code? Any suggestions on what I should
> add?
> > Can I add things like:
> >
> > { "tls-ms-i", "5061"
2009/1/29 Andrew Rechenberg Lists :
> I'm wondering how I should add this code to Sofia. I think I should be
> adding it to nta.c in const sip_dnstports[] and possibly in
> tports_sips[].
>
> Anywhere else I need to add code? Any suggestions on what I should add?
> Can I add things like:
>
> { "t
Good day,
I'd like to add some code to Sofia so that it has the ability to lookup
SRV records for Microsoft Office Communications Servers. Of course MS
doesn't use the standard for SIP TLS (_sips._tcp.example.com) by
default. They created their own DNS records.
Here are the SRV records for a MS
Hello All,
Just getting into the tel URI concept. Would you suppose a phone is
configured to always send a SIP URI or always send a tel URI in INVITES? Or
do you think the phone should send a mix (i.e. either type of URI) based on
a dial plan?
Best Regards,
Jerry
-
Hi,
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ext mark
morreny
Sent: Wednesday, May 14, 2008 5:22 AM
To: sofia-sip-devel@lists.sourceforge.net
Subject: [Sofia-sip-devel] question about writing a sip client
Hi,
I am looking for a sip client that can display streaming video during a
multi-party conference session. I have done some research on Goggle and the
only one I can find is xLite but it does not have the multiple video output
that I am looking for. As a result of that, I am struggling with the
I know that part of their stuff involves a key that I don't believe is
publicly available. If you figure this out, please share.
Mike
On Feb 2, 2008, at 8:26 AM, Loell Erecre wrote:
hi, i was just hoping if the devs have the slightest idea, if
there's any way to discover a modified algorit
hi, i was just hoping if the devs have the slightest idea, if there's any way
to discover a modified algorithm of a digest authentication used by a
proprieatary sip server.
i have been scouring the web on how yahoo sip works then i came across this one,
http://blog.motiwala.com/2007/08/18/yahoo
2007/8/10, Chung Pak Lai <[EMAIL PROTECTED]>:
> I have implemented a SIP server. Right now, I am trying to deal with the
> problem if the SIP client didn't send a BYE and terminated the socket
> connection. What is the best way to deal with it? For TCP SIP connection, is
> there any notification to
2007/8/15, Adi Baron <[EMAIL PROTECTED]>:
> Having the need of supporting multiple udp/tcp/tls ports - Different port
> per each endpoint the gateway supports.
> Can it be implemented using sofia?
Oh yes.
> What would be the way doing it?
You need either to use nta (and use nta_agent_add_tport(
Having the need of supporting multiple udp/tcp/tls ports - Different port per
each endpoint the gateway supports.
Can it be implemented using sofia?
What would be the way doing it?
Thank you in advance
Adi Baron.
--
Hi,
I have implemented a SIP server. Right now, I am trying to deal with the
problem if the SIP client didn't send a BYE and terminated the socket
connection. What is the best way to deal with it? For TCP SIP
connection, is there any notification to let me know if the SIP
connection has been termi
On 11/28/06, Alex Guan <[EMAIL PROTECTED]> wrote:
> Is there a way to use Sofia JUST for its parser?
>
> We are using Sofia in two different applications. In one of the
> applications we would like to parse an incoming character array and generate
> an output of sip_t structure without running NUA
Hi everyone,
Is there a way to use Sofia JUST for its parser?
We are using Sofia in two different applications. In one of the
applications we would like to parse an incoming character array and generate
an output of sip_t structure without running NUA or NTA. I have looked at
the sip_parser.c
On 11/16/06, Pekka Pessi <[EMAIL PROTECTED]> wrote:
> On 11/16/06, Chung Pak Lai <[EMAIL PROTECTED]> wrote:
> > So I would expect it sends a RE-INVITE/UPDATE after 1 second; however,
> > after 1 second, the event callback got an nua_r_invite with status 900
> > and "Internal NUA error" followed by
On 11/16/06, Chung Pak Lai <[EMAIL PROTECTED]> wrote:
> So I would expect it sends a RE-INVITE/UPDATE after 1 second; however,
> after 1 second, the event callback got an nua_r_invite with status 900
> and "Internal NUA error" followed by nua_i_state. Could you see if there
> is anything I am not d
ai
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] Question regarding sip call termination
On 11/14/06, Chung Pak Lai <[EMAIL PROTECTED]> wrote:
> I got a question regarding to the sip call termination in sofia-sip.
> As I am using sofia-sip with TCP connection,
On 11/14/06, Chung Pak Lai <[EMAIL PROTECTED]> wrote:
> I got a question regarding to the sip call termination in sofia-sip. As I am
> using sofia-sip with TCP connection, if the sip call's connection is broken
> in the middle of the call without sending bye message, will sofia-sip detect
> the con
Hello,
I got a question
regarding to the sip call termination in sofia-sip. As I am using sofia-sip
with TCP connection, if the sip call's connection is broken in the middle of the
call without sending bye message, will sofia-sip detect the connection failure,
or it's up to the application
Hi,
On Wed, 10 May 2006, Saurav SAHU wrote:
Which sofia-sip tag does the application use to determine the selected
codec as a result of the SOA media negotiation?
you'll have to check the whole SDP. The tags are SOATAG_USER_SDP()
(parsed) and SOATAG_USER_SDP_STR() (raw text). These get you t
Hi all
Which sofia-sip tag does the
application use to determine the selected codec as a result of
the SOA media negotiation?
In the below scenario I'm making an outgoing call from my sofia-sip
application to X-Lite. The sofia-sip application sends an offer with 2 codecs in
INVITE where
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