In the end, I managed to make the app work.
And to add shame, it was just a matter of a single configuration field,
indeed. After I set srtp_secure_signaling = 0, it started to work.
It's almost embarassing.
Thanks anyway for your time and efforts.
2015-02-11 10:51 GMT+01:00 Harald Radke <harryrat at gmx.de>:
>
> hm, I am really not familiar with TLS and RFC5626, so this is just some
> guesswork:
>
> do you really need RFC5626 in your setup ? if not ( or just for testing),
> I would disable it in your account config and try it again. Also, can you
> compile and test your code on an non IOS device, idally PC or Mac? In case
> of no luck there, too, you could do some wireshark tracing to find out the
> differences in data flows between your PJSIP app and an arbitrary (working)
> SIP UA
>
> Regards,
>
> Harry
> *Gesendet:* Dienstag, 10. Februar 2015 um 13:20 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)
> 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
>>> _______________________________________________ Visit our blog:
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>>>
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>>>
>>
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>>
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>
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