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Old 28th June 2011
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Oko Oko is offline
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Location: Kosovo, Serbia
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Default VoIP on OpenBSD

I got sick and tired of cranky people complaining about the lack of Skype on OpenBSD. So, I just wrote VoIP how-to for OpenBSD users who have dynamica IP address sitting behind NAT (most people on the world). It assumes that you have a free account on Proxy SIP servers




The how-to is in Serbian for now


but I will translate soon. In the mean time this is shorten version. No configuration of OpenBSD is needed unless you want to listen to music while making a phone call. In that case you need to start audio server.

$ aucat -l
No configuration of your router which does NAT and any kind a funny stuff is needed!

The only thing is that pf rules must be very weak since SIP reminds me of FTP. It starts the conversation on UDP 5060 but then depends on the location uses ports all over the place.

You will need to configure files with your account information thought.

$more .pjsua.ekiga
# sample config file for Ekiga, through NAT
similarly for iptel.org

$more .pjsua.iptel
# sample config file for iptel.org, through NAT
Now you can start pjsua client for example with iptel.org account.

pjsua --config-file .pjsua.iptel
You will get output

|       Call Commands:         |   Buddy, IM & Presence:  |     Account:      |
|                              |                          |                   |
|  m  Make new call            | +b  Add new buddy       .| +a  Add new accnt |
|  M  Make multiple calls      | -b  Delete buddy         | -a  Delete accnt. |
|  a  Answer call              |  i  Send IM              | !a  Modify accnt. |
|  h  Hangup call  (ha=all)    |  s  Subscribe presence   | rr  (Re-)register |
|  H  Hold call                |  u  Unsubscribe presence | ru  Unregister    |
|  v  re-inVite (release hold) |  t  ToGgle Online status |  >  Cycle next ac.|
|  U  send UPDATE              |  T  Set online status    |  <  Cycle prev ac.|
| ],[ Select next/prev call    +--------------------------+-------------------+
|  x  Xfer call                |      Media Commands:     |  Status & Config: |
|  X  Xfer with Replaces       |                          |                   |
|  #  Send RFC 2833 DTMF       | cl  List ports           |  d  Dump status   |
|  *  Send DTMF with INFO      | cc  Connect port         | dd  Dump detailed |
| dq  Dump curr. call quality  | cd  Disconnect port      | dc  Dump config   |
|                              |  V  Adjust audio Volume  |  f  Save config   |
|  S  Send arbitrary REQUEST   | Cp  Codec priorities     |  f  Save config   |
|  q  QUIT   L  ReLoad   sleep MS   echo [0|1|txt]     n: detect NAT type     |
You have 0 active call
Now make a test phone call.

>>> m
(You currently have 0 calls)
Buddy list:

   0         For current dialog.
  -1         All 0 buddies in buddy list
  [1 - 0]    Select from buddy list
  URL        An URL
  <Enter>    Empty input (or 'q') to cancel
Make call:
You need to type sip address you are calling. My configuration is tested with the following.

Make call: sip:*010600@ekiga.net
Now you should her a female voice and see something like
t 0 (default)
 23:28:46.948   conference.c  Port 0 (default) transmitting to port 3 (sip:*010600@ekiga.net)
 23:28:46.948    pjsua_app.c  Media for call 1 is active
 23:28:46.948   pjsua_core.c  TX 522 bytes Request msg ACK/cseq=19213 (tdta0x89135000) to UDP
ACK sip:600@ SIP/2.0
Via: SIP/2.0/UDP;rport;branch=z9hG4bKPjREb8mml5UfL1taglP1Oas-27C9cZjJT5
Max-Forwards: 70
From: sip:ppunosevac@ekiga.net;tag=Fodbqp2tIT9wxRTlDe2ewLK-W8MonT77
To: sip:*010600@ekiga.net;tag=as507a14ed
Call-ID: 2ivSEtm.d4G99AYmcLRIS7JbzDJ05c8E
CSeq: 19213 ACK
Route: <sip:;lr;did=90f.5093f8f4>
Route: <sip:;lr;ftag=Fodbqp2tIT9wxRTlDe2ewLK-W8MonT77>
Route: <sip:;lr;ftag=Fodbqp2tIT9wxRTlDe2ewLK-W8MonT77>
Content-Length:  0

--end msg--
 23:28:46.948    pjsua_app.c  Call 1 state changed to CONFIRMED
 23:28:47.584 strm0x89137974  VAD re-enabled
User-Agent: PJSUA v1.6/i386-unknown-openbsd4.8
Content-Type: application/sdp
Content-Length:   445

o=- 3518220550 3518220550 IN IP4
c=IN IP4
t=0 0
m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4001 IN IP4
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Last edited by Oko; 3rd April 2014 at 09:24 PM.
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