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)
why is this? how can it resolve the SIP URI to that IP address?
//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
>> >>
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>> >> pjsip at lists.pjsip.org
>> >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>> >
>> >
>> > _______________________________________________
>> > Visit our blog: http://blog.pjsip.org
>> >
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>> > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
>> >
>> >
>>
>> _______________________________________________
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>>
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>
>
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