Re: [Sofia-sip-devel] performance question

2009-06-22 Thread Michael Jerris 
Are you using nta or nua?

On Jun 22, 2009, at 12:43 AM, jonathan augenstine  
 wrote:

> I am developing a simple redirect server on Windows 2008 Server.   
> When an invite arrives, the handler performs a database lookup, and  
> then returns a 302 response with the database response in the  
> Contact field.  The database access is about 15-30 milliseconds even  
> under high load.  What I am seeing are performance issues I would  
> not expect.  Performance is OK up to about 30 invite messages per  
> second.  Then I start to see significant lag occurring.  What  
> happens is that I see log messages indicating the response has be  
> sent, however, wireshark is indicating that the 302 may not go out  
> until after a significant delay.  At 30 messages per second I see  
> delays of 1-2 seconds.  Not extreme but it is a concern.  When I  
> start running loads up to 50-100 invites per second I start to see  
> large delays (15-30 seconds) from arrival of the invite to return of  
> the 302.  I am confident the delay is not in the database lookup.
>
> I have two questions.  What kind of considerations should I make to  
> optimize performance of the Sofia-SIP stack?  The other question is,  
> can someone provide me with some guidance in how to go about  
> troubleshooting this issue?  I have run out of ideas on isolating  
> the problem.
>
> Jonathan
>
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[Sofia-sip-devel] Retrieve Header Content

2009-06-22 Thread Miguel Goncalves 
Miguel Goncalves wrote:
> hello there..
>
>
> this could seem easy but i just cant see it..
>
> im trying to retrieve and rtsp url that is in a sip invite message 
> header, on content_type section of the header.. like this:
>
>  SIP/2.0 200 OK
>   Via: SIP/2.0/UDP 
> 192.168.0.110:51458;rport=51458;branch=z9hG4bKDrt305Hmycr8H
>   Record-Route: 
>   Record-Route: 
>   From: ;tag=BX8rFa5N6v9BS
>   To: ;tag=1265391275
>   Call-ID: 18988e35-d90e-122c-cca6-001dba24c9ef
>   CSeq: 116677944 INVITE
>   Contact: 
>   Content-Type: message/external-body; access-type="URL"; 
> expiration="Sat, 01 January 2011 09:00:00 GMT"; 
> URL="rtsp://127.0.0.1:5554/test"
>   User-Agent: eXosip/3.1.0
>   Content-Length: 0
>
> im trying to get the content type on the callback of the event 
> nua_r_invite with this:
>
> void app_r_invite(int status, char const *phrase,
>nua_t *nua, cliente *cli,
>nua_handle_t *nh, sip_t const *sip,
>tagi_t tags[])
> {
>
>  printf(": INVITE : %03d %s\n", status, phrase);
>
>
>if (!sip) {
>printf(": NOTIFY sip=%p,nh=%p\n", sip, nh);
>return;
>}
>
>  if ( status == 200 )
>{
>if(sip)
>{
>
>char *header;
>header = sip_header_as_string( cli->home, (void*) 
> sip->sip_content_type);
>  printf("\nCONTENT TYPE -> %s\n", header);
>}
>
> but i just keep getting :
> INVITE : 200 OK
>
> CONTENT TYPE -> (null)
>
> can anyone help please? Where should i get the content_type and how?
>
> Miguel
>


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[Sofia-sip-devel] problems with running sofsip-cli

2009-06-22 Thread Luo Cheng 
Hi,

I was trying to test sofsip-cli between my nokia n810 and laptop, both
laptop and n810 connected to my home AP with private IP. It seems that I can
receive the notice for incoming call, but when i was trying to answer the
incoming call, it's failed.

here is the printout for sofsip-cli

=
: incoming call
From: ;tag=95U5B6DDFNUmB
To:  
Please Answer(a), decline(d) or Decline(D) the call
> sofsip_cli[1497]: GLIB DEBUG default - priv_set_remote_sdp
sofsip_cli[1497]: GLIB DEBUG default - priv_update_tx_elements
sofsip_cli[1497]: GLIB CRITICAL ** default - priv_update_tx_elements:
assertion `self->sm_udpsink != NULL' failed
> a
sofsip_cli[1497]: GLIB DEBUG default - priv_activate_gst
sofsip_cli[1497]: GLIB DEBUG default - priv_static_capabilities_gst
Set local SDP based on capabilities: v=0
m=audio 0 RTP/AVP 0
a=rtpmap:0 PCMU/8000

sofsip_cli[1497]: GLIB DEBUG default - priv_set_local_sdp
sofsip_cli[1497]: GLIB DEBUG default - priv_update_tx_elements
sofsip_cli[1497]: GLIB CRITICAL ** default - priv_update_tx_elements:
assertion `self->sm_udpsink != NULL' failed
sofsip_cli[1497]: GLIB DEBUG default - priv_activate_gst
sofsip_cli[1497]: GLIB DEBUG default - priv_setup_rtpelements
Succesfully bound to local port 16384.
su_source_port_create() returns 0x74040
sofsip_cli[1497]: GLIB DEBUG default - Setting sockfd to 10.
sofsip_cli[1497]: GLIB DEBUG default - Setting aux-sockfd to -1.
sofsip_cli[1497]: GLIB DEBUG default - farsight_netsocket_stun_map
sofsip_cli[1497]: GLIB DEBUG default - server:(null) domain:192.168.0.3.
UA: answering to the offer received from ;tag=95U5B6DDFNUmB
sofsip> sofsip> sofsip_cli[1497]: GLIB DEBUG default - cb_stun_state:
stun_error (8)sofsip_cli[1497]: GLIB DEBUG default - cb_stun_state: error in
communication with the STUN server
sofsip_cli[1497]: GLIB DEBUG default - priv_cb_ready
sofsip_cli: ssc_media_gst.c:563: priv_cb_ready: Assertion `audiosink !=
((void *)0)' failed.


WARNING: The program has received signal (6) and will terminate.
~ $ sh: turning off NDELAY mode
==
Anyone knows where got wrong?

regards
Cheng
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