[Asterisk-Users] SIP to SIP sadness

[BJ Weschke]

 SIP requires RTP connections in addition to the signaling connection which
normally happens on UDP 5060. The RTP connections vary in port usage (the
range is configurable through rtp.conf) and are nearly impossible to get
going without some "man in the middle" help when you have two Asterisk
servers that are both behind NAT firewalls.
  If that's the case, you're much better off here with IAX where your
signaling and media stream can be consolidated into a single stream.

 On 10/17/05, Michael Furdyk <mfurdyk at takingitglobal.org> wrote:
>
> Okay so it seems like it was the firewall, someone just suggested that we
> disable it (On Redhat server) and it's working fine... so does anyone know
> clearly what ports (other than 5060) SIP uses for these calls?
>  -- Mike
>
>  ------------------------------
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Michael Furdyk
> *Sent:* October 17, 2005 2:54 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [Asterisk-Users] SIP to SIP sadness
>
>   Wow, after getting the O'Reilly book delivered last week along with two
> Digium TDM400P's, I'm really getting the hang of this. But the SIP to SIP
> issue is still a problem... and it seems silly because everything else
> (should have been?) so much harder but is working pretty flawlessly.
> Basically I get no audio either way, and it tries to do a "native bridge"
> (handoff?)
>  So when I dial another SIP extension, I get:
>   ---
> -- SIP/324-ab4d answered SIP/322-7e8d
> We're at 192.168.1.195 <http://192.168.1.195/> port 16874
> Answering with preferred capability 0x2 (gsm)
> Answering with preferred capability 0x4 (ulaw)
> Answering with non-codec capability 0x1 (telephone-event)
> Reliably Transmitting (NAT) to 192.168.1.24:5060<http://192.168.1.24:5060/>
> :
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.1.24:5060 <http://192.168.1.24:5060/>
> ;branch=z9hG4bK8E845F95F34044ACA77C54EF28288C32;received=192.168.1.24<http://192.168.1.24/>
> ;rport=5060
> From: Michael Furdyk <sip:322 at 192.168.1.195>;tag=411158625
> To: <sip:324 at 192.168.1.195>;tag=as6606adb1
> Call-ID: 28E78AC1-5FDE-414E-8059-68B393A24F60 at 192.168.1.24
> CSeq: 30931 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: <sip:324 at 192.168.1.195>
> Content-Type: application/sdp
> Content-Length: 239
>  v=0
> o=root 3348 3348 IN IP4 192.168.1.195 <http://192.168.1.195/>
> s=session
> c=IN IP4 192.168.1.195 <http://192.168.1.195/>
> t=0 0
> m=audio 16874 RTP/AVP 3 0 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>  ---
> -- Attempting native bridge of SIP/322-7e8d and SIP/324-ab4d
>  <-- SIP read from 192.168.1.24:5060 <http://192.168.1.24:5060/>:
> ACK sip:324 at 192.168.1.195 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.24:5060 <http://192.168.1.24:5060/>
> ;rport;branch=z9hG4bKB644BADE71EF4422878597A96BE8D613
> From: Michael Furdyk <sip:322 at 192.168.1.195>;tag=411158625
> To: <sip:324 at 192.168.1.195>;tag=as6606adb1
> Contact: <sip:322 at 192.168.1.24:5060>
> Call-ID: 28E78AC1-5FDE-414E-8059-68B393A24F60 at 192.168.1.24
> CSeq: 30931 ACK
> Max-Forwards: 70
> Content-Length: 0
>  Here is my default in SIP.conf. Each SIP config has canreinvite=no
>  [general]
> disallow=all
> allow=gsm
> allow=ulaw
> nat=no
> canreinvite=no
> externip=(real external IP is here)
> localnet=192.168.1.195/255.255.255.0 <http://192.168.1.195/255.255.255.0>
> srvlookup=yes
> sipdebug=yes
> I have tried nat=no and nat=yes
>
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