On Mon, Jul 14, 2008 at 2:05 PM, Zahed Emon <ena2413 at gmail.com> wrote:
> hi again,
>
> >>Have you look at the log?
>
> yes; I have , the to , from and contact look OK (as desired).
> but when I looked into the destination IP address of the invite in
> wireshark I got this IP 64.14.244.60.
>
> this is because the DNS resolver is not available and then it resolves the
> sip URI to that IP Address by gethostname(). (from the log below in red)
>
Well, yes.
>
> why is this? how can it resolve the SIP URI to that IP address?
>
I'm not sure I understand the question. If you specify a hostname in the
URI, do you expect it to do something else other than to resolve it to an IP
address?
Cheers
benny
>
> //EMON
>
> here is some snap from log file..
>
> 14:43:22.949 tsx01752D0C Timeout timer event
> 14:43:22.949 tsx01752D0C State changed from Calling to Terminated,
> event=TIMER
> 14:43:22.949 pjsua_acc.c SIP registration failed, status=408 (Request
> Timeout)
> 14:43:22.952 tsx01752D0C Timeout timer event
> 14:43:22.952 tsx01752D0C State changed from Terminated to Destroyed,
> event=TIMER
> 14:43:22.952 tdta01751C78 Destroying txdata Request msg
> REGISTER/cseq=43937 (tdta01751C78)
> 14:43:22.952 tsx01752D0C Transaction destroyed!
> 14:44:16.234 pjsua_media.c pjsua_set_snd_dev(): attempting to open devices
> @16000 Hz
> 14:44:16.408 pasound.c Opened device Microphone (Realtek High
> Defini(MME)/Speakers (Realtek High Definiti(MME) for recording and playback,
> sample rate=16000, ch=1, bits=16, 320 samples per frame, input latency=100
> ms, output latency=100 ms
> 14:44:16.408 pasound.c Starting Microphone (Realtek High Defini
> stream..
> 14:44:16.409 pasound.c PA message: Pa_StartStream: waveInStart
> returned = 0x0.
>
> 14:44:16.409 pasound.c Done, status=0
> 14:44:16.410 echo_speex.c Speex Echo canceller/AEC created,
> clock_rate=16000, samples per frame=320, tail length=200 ms, latency=200 ms
> 14:44:16.412 pjsua_call.c Making call with acc #2 to sip:zahed2 at xxx.com<sip%3Azahed2 at xxx.com>
> 14:44:16.412 dlg02C7817C UAC dialog created
> 14:44:16.420 pjsua_media.c Media index 0 selected for call 0
> 14:44:16.421 dlg02C7817C Module mod-invite added as dialog usage,
> data=01738FB8
> 14:44:16.422 dlg02C7817C Session count inc to 2 by mod-invite
> 14:44:16.422 dlg02C7817C Module mod-100rel added as dialog usage,
> data=02C78EB4
> 14:44:16.422 dlg02C7817C 100rel module attached
> 14:44:16.422 inv02C7817C UAC invite session created for dialog
> dlg02C7817C
> 14:44:16.422 endpoint Request msg INVITE/cseq=15363 (tdta02C7CE88)
> created.
> 14:44:16.422 inv02C7817C Sending Request msg INVITE/cseq=15363
> (tdta02C7CE88)
> 14:44:16.422 dlg02C7817C Sending Request msg INVITE/cseq=15363
> (tdta02C7CE88)
> 14:44:16.422 tsx0177F4E4 Transaction created for Request msg
> INVITE/cseq=15362 (tdta02C7CE88)
> 14:44:16.422 tsx0177F4E4 Sending Request msg INVITE/cseq=15362
> (tdta02C7CE88) in state Null
> 14:44:16.422 sip_resolve.c DNS resolver not available, target 'xxx.com:0'
> type=Unspecified will be resolved with gethostbyname()
> 14:44:16.461 pasound.c Recorder thread started
> 14:44:16.462 pasound.c Player thread started
> 14:44:16.462 aec01751CCC AEC reset, delay=0, prefetch=4
> 14:44:16.462 aec01751CCC AEC Info: old frame removed (seq=1, want=-2,
> count=1)
> 14:44:16.463 aec01751CCC AEC Info: empty queue for seq=-2!
> 14:44:16.463 aec01751CCC AEC reset, delay=-2, prefetch=4
> 14:44:16.515 aec01751CCC AEC reset, delay=-3, prefetch=4
> 14:44:16.515 aec01751CCC AEC Info: old frame removed (seq=2, want=-2,
> count=1)
> 14:44:16.515 aec01751CCC AEC Info: empty queue for seq=-2!
> 14:44:16.568 aec01751CCC AEC Info: old frame removed (seq=3, want=-1,
> count=1)
> 14:44:16.568 aec01751CCC AEC Info: empty queue for seq=-1!
> 14:44:16.568 aec01751CCC AEC reset, delay=-4, prefetch=4
> 14:44:16.568 aec01751CCC AEC Info: prefetching (first seq=4)
> 14:44:16.621 aec01751CCC AEC Info: prefetching (first seq=4)
> 14:44:16.621 aec01751CCC AEC Info: prefetching (first seq=4)
> 14:44:16.675 aec01751CCC AEC Info: prefetching (first seq=4)
> 14:44:16.675 aec01751CCC AEC info: queue is full, frame discarded
> [count=6, seq=10]
> 14:44:16.675 aec01751CCC AEC reset, delay=-1, prefetch=4
> 14:44:16.675 aec01751CCC AEC Info: prefetching (first seq=10)
> 14:44:16.675 aec01751CCC AEC Info: prefetching (first seq=10)
> 14:44:16.728 aec01751CCC AEC Info: prefetching (first seq=10)
> 14:44:16.750 sip_resolve.c Target 'wipe.xxx:0' type=Unspecified resolved
> to '64.34.46.60:5060' type=UDP (UDP transport)
> 14:44:16.751 pjsua_core.c TX 1045 bytes Request msg INVITE/cseq=15362
> (tdta02C7CE88) to UDP 64.34.46.60:5060:
> INVITE sip:zahed2 at xxx.com <sip%3Azahed2 at xxx.com> SIP/2.0
> Via: SIP/2.0/UDP 92.254.195.215:5060
> ;rport;branch=z9hG4bKPj13ac3b662210476abf55442c4f222d98
> Max-Forwards: 70
> From: sip:zahed1@xxxxxxx <sip%3Azahed1 at xxx.com>
> ;tag=649c0a49fcb7456582fb8b58e9180fa0
> To: sip:zahed2 at xxx.com <sip%3Azahed2 at xxx.com>
> Contact: <sip:zahed1 at 92.254.195.215:5060>
> Call-ID: e9e69b640f504afdba1121ee6e8fec4a
> CSeq: 15362 INVITE
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH,
> REFER, MESSAGE, OPTIONS
> Supported: replaces, 100rel, norefersub
> User-Agent: PJSUA v0.9.0-release/win32
> Content-Type: application/sdp
> Content-Length: 465
>
> v=0
> o=- 3425035456 3425035456 IN IP4 92.254.195.215
> s=pjmedia
> c=IN IP4 92.254.195.215
> t=0 0
> a=X-nat:0
> m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 101
> a=rtcp:4001 IN IP4 92.254.195.215
> a=rtpmap:103 speex/16000
> a=rtpmap:102 speex/8000
> a=rtpmap:104 speex/32000
> a=rtpmap:117 iLBC/8000
> a=fmtp:117 mode=30
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a=sendrecv
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
>
>
>
>
> On Mon, Jul 14, 2008 at 12:03 PM, Benny Prijono <bennylp at pjsip.org> wrote:
> > On Mon, Jul 14, 2008 at 10:45 AM, Zahed Emon <ena2413 at gmail.com> wrote:
> >>
> >> Hi,
> >>
> >> yes, I also think that the INVITE will be send directly. BUt I am not
> >> providing any URI which has that IP. In my network that IP is not
> >> available. I dont know from where it is getting the IP.
> >
> > Have you look at the log?
> >
> >>
> >> is this any default stun/turn server's IP?
> >>
> >
> > No. SIP requests are sent to the target URI or Route URI, and that's just
> > about it.
> >
> > Cheers
> > Benny
> >
> >
> >>
> >> BR
> >>
> >> Zahed
> >>
> >> On Mon, Jul 14, 2008 at 11:00 AM, Benny Prijono <bennylp at pjsip.org>
> wrote:
> >> > On Sat, Jul 12, 2008 at 9:11 PM, Zahed Emon <ena2413 at gmail.com>
> wrote:
> >> >>
> >> >> Hi,
> >> >>
> >> >> I have created my small app which just register with SIP server and
> >> >> makes call to other client.
> >> >>
> >> >> now the sip registration works fine. but I when tried to call other
> >> >> registered user I saw not SIP invite is going to SIP server rather
> >> >> they goes to some IP like 68.44.244.60. eventually it does not get
> any
> >> >> response and dies.
> >> >>
> >> >> I dont know why it is sending invite to the IP address? I tried with
> >> >> the pjsua_app application that comes with the distribution and still
> >> >> have the same problem.
> >> >>
> >> >
> >> > Since there's no route set, the INVITE will be sent to the target URI
> >> > (the
> >> > "URI" variable in your snippet). Is this not the case?
> >> >
> >> > -benny
> >> >
> >> >
> >> >>
> >> >> I am using UDP transport for SIP. I am behind a firewall that stops
> >> >> every UDP packet send outside and coming inside.
> >> >>
> >> >> please help me solve this problem.
> >> >>
> >> >> here is some snap of the code
> >> >>
> >> >> //Emon
> >> >>
> >> >> //add UDP tranport
> >> >>
> >> >> pjsua_transport_config trans_cfg_UDP;
> >> >> pjsua_transport_config_default(&trans_cfg_UDP);
> >> >> trans_cfg.public_addr = pj_str("xxx.yyy.zzz.146");
> >> >>
> >> >> status = pjsua_transport_create(PJSIP_TRANSPORT_UDP,
> >> >> &trans_cfg_UDP,
> >> >> &trans_id_UDP);
> >> >> if(status != PJ_SUCCESS)
> >> >> {
> >> >> error_exit("Error in creating UDP transport", status);
> >> >> }
> >> >>
> >> >>
> >> >> acc_cfg.id = pj_str(uri);
> >> >> acc_cfg.reg_uri = pj_str ("sip:"MYdomain);
> >> >> acc_cfg.cred_count = 1;
> >> >> acc_cfg.cred_info[0].realm = pj_str("*");
> >> >> acc_cfg.cred_info[0].scheme = pj_str("digest");
> >> >> acc_cfg.cred_info[0].username = pj_str(user);
> >> >> acc_cfg.cred_info[0].data_type =
> >> >> PJSIP_CRED_DATA_PLAIN_PASSWD;
> >> >> acc_cfg.cred_info[0].data = pj_str(pass);
> >> >> }
> >> >>
> >> >>
> >> >> status = pjsua_acc_add(&acc_cfg,PJ_TRUE,&acc_id);
> >> >> if(status != PJ_SUCCESS)
> >> >> {
> >> >> error_exit("Error in registration", status);
> >> >> }
> >> >>
> >> >> pj_str_t uri = pj_str(URI);
> >> >>
> >> >> status = pjsua_call_make_call(acc_id,&uri,0,NULL,NULL,NULL);
> >> >> if(status != PJ_SUCCESS)
> >> >> {
> >> >> error_exit("Error in Making call", status);
> >> >> }
> >> >>
> >> >> _______________________________________________
> >> >> Visit our blog: http://blog.pjsip.org
> >> >>
> >> >> pjsip mailing list
> >> >> pjsip at lists.pjsip.org
> >> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> >> >
> >> >
> >> > _______________________________________________
> >> > Visit our blog: http://blog.pjsip.org
> >> >
> >> > pjsip mailing list
> >> > pjsip at lists.pjsip.org
> >> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> Visit our blog: http://blog.pjsip.org
> >>
> >> pjsip mailing list
> >> pjsip at lists.pjsip.org
> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> >
> >
> > _______________________________________________
> > Visit our blog: http://blog.pjsip.org
> >
> > pjsip mailing list
> > pjsip at lists.pjsip.org
> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
> >
> >
>
>
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
> pjsip mailing list
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>
>
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