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: >>> 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 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 > > -------------- next part -------------- An HTML attachment was scrubbed... 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