Re: [Sofia-sip-devel] performance question
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 > > --- > --- > --- > - > Are you an open source citizen? Join us for the Open Source Bridge > conference! > Portland, OR, June 17-19. Two days of sessions, one day of > unconference: $250. > Need another reason to go? 24-hour hacker lounge. Register today! > http://ad.doubleclick.net/clk;215844324;13503038;v?http://opensourcebridge.org > ___ > Sofia-sip-devel mailing list > Sofia-sip-devel@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel -- Are you an open source citizen? Join us for the Open Source Bridge conference! Portland, OR, June 17-19. Two days of sessions, one day of unconference: $250. Need another reason to go? 24-hour hacker lounge. Register today! http://ad.doubleclick.net/clk;215844324;13503038;v?http://opensourcebridge.org ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] Retrieve Header Content
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 > -- Are you an open source citizen? Join us for the Open Source Bridge conference! Portland, OR, June 17-19. Two days of sessions, one day of unconference: $250. Need another reason to go? 24-hour hacker lounge. Register today! http://ad.doubleclick.net/clk;215844324;13503038;v?http://opensourcebridge.org ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] problems with running sofsip-cli
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 -- Are you an open source citizen? Join us for the Open Source Bridge conference! Portland, OR, June 17-19. Two days of sessions, one day of unconference: $250. Need another reason to go? 24-hour hacker lounge. Register today! http://ad.doubleclick.net/clk;215844324;13503038;v?http://opensourcebridge.org___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel