Re: Antwort: [Siproxd-users] siproxd zyxel2000W problems

[Peter K. <pe...@as...>]

I have just tcpdumped a complete inbound call and discovered that
the local phone in fact does not send out a single RTP packet... no wonder
I don't hear anything. very strange... so I guess there must be something
wrong with the RTP information that gets to the phone on initiating the
call when picking up the phone. The sipproxy does open both rtpproxy
streams, so there must be something wrong with the message itself...

this is the OK message sent from the proxy to the phone, with 9999999 the
sip phone number, 493012345678 the land-line phone calling, 
and 192.158.99.99 the external ip number of the local net:
---------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK15a2.924f6011.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK15a2.f942f8.0
Via: SIP/2.0/UDP 195.226.161.164:5060;branch=z9hG4bK3a883ae7
Record-Route: <sip:9999999@217.10.79.9;ftag=as7fff67b2;lr=on>
Record-Route: <sip:4930869999999@217.10.79.8;ftag=as7fff67b2;lr=on>
From: "03012345678" <sip:03012345678@195.226.161.164>;tag=as7fff67b2
To: <sip:493...@si...>;tag=4D1DA4377F1766B19EF
Call-ID: 24292372190406cf552636d14167163d@195.226.161.164
CSeq: 102 INVITE
Contact: <sip:4930869999999@192.168.1.125:5060;transport=udp>
user-agent: ZyXEL P2000W VoIP Wi-Fi Phone
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, SUBSCRIBE, NOTIFY, INFO, REFER
Content-Type: application/sdp
Content-Length:   151

v=0
o=TelogyUnknown0000 59871 59871 IN IP4 195.158.99.99
s=RTP Audio
c=IN IP4 195.158.99.99
t=0 0
m=audio 7070 RTP/AVP 8
a=rtpmap:8 PCMA/8000
---------------------------------------------------------------------------



tcpdump shows me LOTS of UDP traffic on 7070 from the proxy to the phone, 
but absolutely nothing from the phone to the proxy. When I call
10000, I get bidirectional traffic...

I am not too familiar with SDP - Any wild guesses about this one? The only 
item I would find questionable would be the RTP format type. When calling
sipgates 10000, I get:

m=audio 16118 RTP/AVP 8 0 3 10 97 18 2 5
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=silenceSupp:off - - - -

sent from sipgate... and when I look at the phones voice codec list,
I see:

G.729 8k
G.711u 64k
G.711a 64k

Does this mean that the phone is not able to provide the right 
encoding for the former request since it got too few choices offered, so 
decides not to send anything? Would be weird, so perhaps not...


-- 
peter koellner <pe...@as...>