Hi,
may I was not clear while asking the question.
when I want to call one URI and I want my client to contact with a
particular sip server to get the contact information about other endpoint.
What should I do?
-- specifiy the proxy sip server with lr
or do in need to do some thing more?
now if I specify the proxy sip server, why should my sip client tries to
resolve the URI to a IP address?
BR
EMON
On Mon, Jul 14, 2008 at 3:33 PM, Benny Prijono <bennylp at pjsip.org> wrote:
> 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
>> >
>> >
>>
>>
>> _______________________________________________
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>>
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>>
>>
>
> _______________________________________________
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>
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