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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