[asterisk-users] one way rtp stream (Sent alwax to 127.0.0.1)

["Hans-Jürgen Brand"]

Found problem

xlite client Version 3 Build 34025 send wrong rtp port to asterisk. But I don't know how to change this at xlite????


venus*CLI>
<-- SIP read from 192.168.100.20:60726:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK02d4cc64;rport=5060
Contact: <sip:xlite at 192.168.100.20:60726;rinstance=45385da6efafa3ea>
To: <sip:xlite at 192.168.100.20:60726;rinstance=45385da6efafa3ea>;tag=7b512144
From: "Hans-Juergen Brand"<sip:snom at 192.168.100.32>;tag=as4530bf3b
Call-ID: 4a11930c07e4aa9256e04885453d8f4d at 192.168.100.32
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 179

v=0
o=- 5 2 IN IP4 127.0.0.1
s=CounterPath X-Lite 3.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 59050 RTP/AVP 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 127.0.0.1:59050
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:xlite at 192.168.100.20:60726;rinstance=45385da6efafa3ea>
set_destination: Parsing <sip:xlite at 192.168.100.20:60726;rinstance=45385da6efafa3ea> for address/port to send to
set_destination: set destination to 192.168.100.20, port 60726
Transmitting (no NAT) to 192.168.100.20:60726:
ACK sip:xlite at 192.168.100.20:60726;rinstance=45385da6efafa3ea SIP/2.0
Via: SIP/2.0/UDP 192.168.100.32:5060;branch=z9hG4bK42575a4c;rport
From: "Hans-Juergen Brand" <sip:snom at 192.168.100.32>;tag=as4530bf3b
To: <sip:xlite at 192.168.100.20:60726;rinstance=45385da6efafa3ea>;tag=7b512144
Contact: <sip:snom at 192.168.100.32>
Call-ID: 4a11930c07e4aa9256e04885453d8f4d at 192.168.100.32
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
                  


-------- Original-Nachricht --------
Datum: Thu, 28 Dec 2006 22:30:24 +0100
Von: "Hans-Jürgen Brand" <hans-juergen.brand at gmx.net>
An: asterisk-users at lists.digium.com
Betreff: [asterisk-users] one way rtp stream (Sent alwax to 127.0.0.1)

> Asterisk version 1.2.14
> 
> I use snom190 and xliteV3 as sip phones.
> asterisk send the rtp stream never to the xlite softphone.
> 
> Any hits for me?
> 
> *CLI> rtp debug
> RTP Debugging Enabled
>     -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack
>     -- Called snom
>     -- SIP/snom-00797110 is ringing
>     -- SIP/snom-00797110 is ringing
>     -- SIP/snom-00797110 answered SIP/xlite-007918f0
>     -- Attempting native bridge of SIP/xlite-007918f0 and
> SIP/snom-00797110
> Got RTP packet from 192.168.100.70:50002 (type 0, seq 6022, ts 32652224,
> len 160)
> Sent RTP packet to 127.0.0.1:17548 (type 0, seq 49874, ts 64, len 160)
> Got RTP packet from 192.168.100.20:17548 (type 0, seq 6911, ts 1973300,
> len 160)Sent RTP packet to 192.168.100.70:50002 (type 0, seq 28956, ts 16,
> len 160)
> Got RTP packet from 192.168.100.70:50002 (type 0, seq 6023, ts 32652544,
> len 160)
> Sent RTP packet to 127.0.0.1:17548 (type 0, seq 49875, ts 384, len 160)   
>    
> 
> 
> 
> 
> *CLI> sip show peers
> Name/username              Host            Dyn Nat ACL Port     Status
> snom/snom                  192.168.100.70   D          2051    
> Unmonitored
> xlite/xlite                192.168.100.20   D          11420   
> Unmonitored
> 2 sip peers [2 online , 0 offline]                    
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