Sorry for the late answer.
Anyway, I just tested again with CSipSimple, with the very same
configuration parameters I'm using on the iOS app and I can both make and
receive calls with the former but not with the latter.
I'm sure is some flag or option I'm forgetting (or don't know about), but
documentation is not helping...
Alberto
2015-02-04 13:59 GMT+01:00 Harald Radke <harryrat at gmx.de>:
> Well, the 407 is not really an error, but the proxy telling your client to
> authenticate itself, which it then does on the second invite, normal
> challange-response
> behaviour. However, I don't actually know, how the proxy reacts on a wrong
> authentication (if it should send you another 407 on wrong auth. or an 200
> OK on correct one, or directly a 1xx ringing)
>
> Sure you can call your destination with another sip phone?
>
> Harry
>
> *Gesendet:* Mittwoch, 04. Februar 2015 um 12:52 Uhr
> *Von:* "Alberto Bitto" <alberto.bitto at gmail.com>
> *An:* "pjsip list" <pjsip at lists.pjsip.org>
> *Betreff:* Re: [pjsip] SIP outbound status for account is not active (iOS
> App)
> After I initiate the call, basically the INVITE message never gets
> answered. This is what happens:
>
> =======
> 2015-02-04 12:46:11.187 pjsip-sample[272:75379] calling
> sips:5183 at 10.213.1.14...
> 12:46:11.188 pjsua_call.c !Making call with acc #0 to
> sips:5183 at 10.213.1.14
> 12:46:11.188 pjsua_aud.c .Set sound device: capture=-1, playback=-2
> 12:46:11.188 pjsua_aud.c ..Opening sound device PCM at 44100/1/20ms
> 12:46:11.188 coreaudio_dev. ...Using VoiceProcessingIO audio unit
> 12:46:11.903 coreaudio_dev. ...core audio stream started
> 12:46:11.912 pjsua_media.c .Call 0: initializing media..
> 12:46:11.913 pjsua_media.c ..RTP socket reachable at
> 192.168.213.101:4000
> 12:46:11.913 pjsua_media.c ..RTCP socket reachable at
> 192.168.213.101:4001
> 12:46:11.913 pjsua_media.c ..Media index 0 selected for audio call 0
> 12:46:11.913 endpoint ...Automatic switch to TLS transport as
> request-URI uses sips scheme.
> 12:46:11.914 endpoint ....Automatic switch to TLS transport as
> request-URI uses sips scheme.
> 12:46:11.914 pjsua_core.c ....TX 1277 bytes Request msg
> INVITE/cseq=15364 (tdta0x16166600) to TLS 172.28.1.41:5061:
> INVITE sips:5183 at 10.213.1.14 SIP/2.0
>
> Via: SIP/2.0/TLS 172.28.1.214:53518
> ;rport;branch=z9hG4bKPjQ.K9S5gjeuCZthDitssbnbxtb9xMiCOm;alias
>
> Max-Forwards: 70
>
> From: sip:test@172.28.1.41;tag=mdFe8C4gMVf.5cqbLic64HMwfCtnStb0
>
> To: sips:5183 at 10.213.1.14
>
> Contact: <sip:test at 172.28.1.214:53518;transport=TLS;ob>
>
> Call-ID: W5GOmefU2BL5D-cyelRoVYmAjnRvp2ma
>
> CSeq: 15364 INVITE
>
> Route: <sip:172.28.1.41:5061;transport=tls;lr>
>
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
> REFER, MESSAGE, OPTIONS
>
> Supported: replaces, 100rel, timer, norefersub
>
> Session-Expires: 1800
>
> Min-SE: 90
>
> Content-Type: application/sdp
>
> Content-Length: 626
>
>
>
> v=0
>
> o=- 3632039171 3632039171 IN IP4 192.168.213.101
>
> s=pjmedia
>
> b=AS:84
>
> t=0 0
>
> a=X-nat:0
>
> m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 96
>
> c=IN IP4 192.168.213.101
>
> b=TIAS:64000
>
> a=rtcp:4001 IN IP4 192.168.213.101
>
> a=sendrecv
>
> a=rtpmap:98 speex/16000
>
> a=rtpmap:97 speex/8000
>
> a=rtpmap:99 speex/32000
>
> a=rtpmap:104 iLBC/8000
>
> a=fmtp:104 mode=30
>
> a=rtpmap:3 GSM/8000
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:96 telephone-event/8000
>
> a=fmtp:96 0-16
>
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:Kkzy4fVdOmSixsFchDZJGvvBZshk/A2ruy5Yj5U8
>
> a=crypto:2 AES_CM_128_HMAC_SHA1_32
> inline:6WKeEzXIn7U7Jlx7M2NI/Yaa+we/TSrmuh47gcwZ
>
>
> --end msg--
> 12:46:11.926 os_core_unix.c Info: possibly re-registering existing thread
> 12:46:11.943 ViewController .......Call 0 state=CALLING
> 12:46:12.068 pjsua_core.c .RX 496 bytes Response msg
> 407/INVITE/cseq=15364 (rdata0x161391d8) from TLS 172.28.1.41:5061:
> SIP/2.0 407 Proxy Authentication Required
>
> Via: SIP/2.0/TLS 172.28.1.214:53518
> ;rport=53518;branch=z9hG4bKPjQ.K9S5gjeuCZthDitssbnbxtb9xMiCOm;alias
>
> From: sip:test@172.28.1.41;tag=mdFe8C4gMVf.5cqbLic64HMwfCtnStb0
>
> To: sips:5183 at 10.213.1.14;tag=b27e1a1d33761e85846fc98f5f3a7e58.1da7
>
> Call-ID: W5GOmefU2BL5D-cyelRoVYmAjnRvp2ma
>
> CSeq: 15364 INVITE
>
> Proxy-Authenticate: Digest realm="172.28.1.41",
> nonce="VNIHr1TSBoNr+jHb3AuhMSRBZoTBWrLI"
>
> Server: kamailio (4.1.6 (x86_64/linux))
>
> Content-Length: 0
>
>
>
>
> --end msg--
> 12:46:12.068 pjsua_core.c ..TX 415 bytes Request msg ACK/cseq=15364
> (tdta0x16964e00) to TLS 172.28.1.41:5061:
> ACK sips:5183 at 10.213.1.14 SIP/2.0
>
> Via: SIP/2.0/TLS 172.28.1.214:53518
> ;rport;branch=z9hG4bKPjQ.K9S5gjeuCZthDitssbnbxtb9xMiCOm;alias
>
> Max-Forwards: 70
>
> From: sip:test@172.28.1.41;tag=mdFe8C4gMVf.5cqbLic64HMwfCtnStb0
>
> To: sips:5183 at 10.213.1.14;tag=b27e1a1d33761e85846fc98f5f3a7e58.1da7
>
> Call-ID: W5GOmefU2BL5D-cyelRoVYmAjnRvp2ma
>
> CSeq: 15364 ACK
>
> Route: <sip:172.28.1.41:5061;transport=tls;lr>
>
> Content-Length: 0
>
>
>
>
> --end msg--
> 12:46:12.069 endpoint ......Automatic switch to TLS transport as
> request-URI uses sips scheme.
> 12:46:12.069 endpoint .......Automatic switch to TLS transport as
> request-URI uses sips scheme.
> 12:46:12.069 pjsua_core.c .......TX 1459 bytes Request msg
> INVITE/cseq=15365 (tdta0x16166600) to TLS 172.28.1.41:5061:
> INVITE sips:5183 at 10.213.1.14 SIP/2.0
>
> Via: SIP/2.0/TLS 172.28.1.214:53518
> ;rport;branch=z9hG4bKPjSM.BFZ.FYxg8Hba9u1UEh02fBS5bIXkw;alias
>
> Max-Forwards: 70
>
> From: sip:test@172.28.1.41;tag=mdFe8C4gMVf.5cqbLic64HMwfCtnStb0
>
> To: sips:5183 at 10.213.1.14
>
> Contact: <sip:test at 172.28.1.214:53518;transport=TLS;ob>
>
> Call-ID: W5GOmefU2BL5D-cyelRoVYmAjnRvp2ma
>
> CSeq: 15365 INVITE
>
> Route: <sip:172.28.1.41:5061;transport=tls;lr>
>
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
> REFER, MESSAGE, OPTIONS
>
> Supported: replaces, 100rel, timer, norefersub
>
> Session-Expires: 1800
>
> Min-SE: 90
>
> Proxy-Authorization: Digest username="test", realm="172.28.1.41",
> nonce="VNIHr1TSBoNr+jHb3AuhMSRBZoTBWrLI", uri="sips:5183 at 10.213.1.14",
> response="a1549883a388c4d5d72829f39142c20a"
>
> Content-Type: application/sdp
>
> Content-Length: 626
>
>
>
> v=0
>
> o=- 3632039171 3632039171 IN IP4 192.168.213.101
>
> s=pjmedia
>
> b=AS:84
>
> t=0 0
>
> a=X-nat:0
>
> m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 96
>
> c=IN IP4 192.168.213.101
>
> b=TIAS:64000
>
> a=rtcp:4001 IN IP4 192.168.213.101
>
> a=sendrecv
>
> a=rtpmap:98 speex/16000
>
> a=rtpmap:97 speex/8000
>
> a=rtpmap:99 speex/32000
>
> a=rtpmap:104 iLBC/8000
>
> a=fmtp:104 mode=30
>
> a=rtpmap:3 GSM/8000
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:96 telephone-event/8000
>
> a=fmtp:96 0-16
>
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:Kkzy4fVdOmSixsFchDZJGvvBZshk/A2ruy5Yj5U8
>
> a=crypto:2 AES_CM_128_HMAC_SHA1_32
> inline:6WKeEzXIn7U7Jlx7M2NI/Yaa+we/TSrmuh47gcwZ
>
>
> --end msg--
> 12:46:12.073 pjsua_core.c .RX 373 bytes Response msg
> 100/INVITE/cseq=15365 (rdata0x161391d8) from TLS 172.28.1.41:5061:
> SIP/2.0 100 trying -- your call is important to us
>
> Via: SIP/2.0/TLS 172.28.1.214:53518
> ;rport=53518;branch=z9hG4bKPjSM.BFZ.FYxg8Hba9u1UEh02fBS5bIXkw;alias
>
> From: sip:test@172.28.1.41;tag=mdFe8C4gMVf.5cqbLic64HMwfCtnStb0
>
> To: sips:5183 at 10.213.1.14
>
> Call-ID: W5GOmefU2BL5D-cyelRoVYmAjnRvp2ma
>
> CSeq: 15365 INVITE
>
> Server: kamailio (4.1.6 (x86_64/linux))
>
> Content-Length: 0
>
>
>
>
> --end msg--
> =======
>
> And after a while, i receive a 408 Timeout error and pjsua disconnects.
>
> I noticed the 407 Proxy Authentication Required error but I can't
> understand it, since using other SIP applications (like CSipSimple) I can
> easily make calls through the same kamailio server using the same
> configuration (TLS and all), even to my iOS app.
>
> 2015-02-04 12:32 GMT+01:00 Harald Radke <harryrat at gmx.de>:
>>
>> I dont think that this debug message is unrelated to your
>> problem...this outbound thingy has something to do with rfc5626 which I
>> guess is not relevant in this case (I think)...
>>
>> Hm, I guess it would be interesting to see the log/debug output after
>> initiating an outgoing call
>>
>> Regards,
>>
>> Harry
>> *Gesendet:* Dienstag, 03. Februar 2015 um 17:23 Uhr
>> *Von:* "Alberto Bitto" <alberto.bitto at gmail.com>
>> *An:* "pjsip list" <pjsip at lists.pjsip.org>
>> *Betreff:* [pjsip] SIP outbound status for account is not active (iOS
>> App)
>> Hi,
>>
>> I'm having problems making calls with the iOS app I'm developing and I
>> can't understand what's wrong.
>>
>> So far I've configured an account that successfully registers on a
>> Kamailio server using TLS transport and receives calls but can't make them.
>>
>> When the account configuration runs, I noticed this debug message (is not
>> marked as error):
>>
>> *pjsua_acc.c ....SIP outbound status for acc 0 is not active*
>>
>> I guess this means I can't send messages and calls? How do I make it
>> active?
>>
>> This is how I'm configuring the account:
>>
>> pjsua_acc_config cfg;
>> pjsua_acc_config_default(&cfg);
>>
>> cfg.id = pj_str("sip:" SIP_USER "@" SIP_URL);
>> cfg.reg_uri = pj_str("sip:" SIP_URL ";transport=tls");
>>
>> cfg.cred_count = 1;
>> cfg.cred_info[0].realm = pj_str(SIP_REALM);
>> cfg.cred_info[0].scheme = pj_str("digest");
>> cfg.cred_info[0].username = pj_str(SIP_USER);
>> cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
>> cfg.cred_info[0].data = pj_str(SIP_PASSWD);
>>
>> cfg.use_srtp = PJMEDIA_SRTP_OPTIONAL;
>> cfg.srtp_secure_signaling = 1;
>>
>> cfg.proxy_cnt = 1;
>> cfg.proxy[0] = pj_str("sip:" SIP_URL ":" SIP_PORT
>> ";transport=tls");
>>
>> cfg.reg_use_proxy = 1;
>>
>> status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
>>
>> What am I missing?
>>
>> Thanks,
>>
>> Alberto
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>
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