[Asterisk-Dev] Help on LeadTek's Latest VideoPhone

Dean Huang ubinh at yahoo.com
Mon Jul 5 08:21:46 MST 2004


Hello All,

I need some helps on LeadTek's latest VideoPhone.

1) Voice functions of the phone works perfect
2) Asterisk always dropped the video part in SDP from one phone and 
invited  the  another phone with only voice capacity

The following is the log from *.

Regards,
D.H.

======================================================================
Jul  5 09:08:29 VERBOSE[1133742896]: 8 headers, 0 lines
Jul  5 09:08:29 DEBUG[1133742896]: Stopping retransmission on 
'50fab160-40794-c0a81dd8 at sipusers.com' of Response 100: Found
Jul  5 09:08:29 VERBOSE[1133742896]:

Sip read:
INVITE sip:9722352288 at sipusers.com:5060;user=phone SIP/2.0

Call-ID: 50fab160-40794-c0a81dd8 at sipusers.com

From: <sip:9722352287 at sipusers.com;user=phone>;tag=20f8c-176cb1

To: <sip:9722352288 at sipusers.com;user=phone>

CSeq: 101 INVITE

Via: SIP/2.0/UDP 192.168.10.19:5061

Contact: <sip:9722352287 at 192.168.10.19:5061;user=phone>

Max-Forwards: 70

Supported: timer

Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS

Expires: 90

Session-Expires: 180

Proxy-Authorization: Digest 
username="9722352287",realm="asterisk",uri="sip:9722352288 at sipusers.com",response="961a05c280b881d57abc6e05043bd523",nonce="58b80f5a"

Content-Type: application/sdp

Content-Length: 246



v=0

o=9722352287 0 0 IN IP4 192.168.10.19

s=-

c=IN IP4 192.168.10.19

t=0 0

m=audio 9838 RTP/AVP 4 0 8 101

a=ptime:30

a=rtpmap:101 telephone-events/8000

m=video 9840 RTP/AVP 34 31

b=AS:128

a=rtpmap:34 H263/90000

a=rtpmap:31 H261/90000


Jul  5 09:08:29 VERBOSE[1133742896]: 15 headers, 12 lines
Jul  5 09:08:29 VERBOSE[1133742896]: Using latest request as basis request
Jul  5 09:08:29 VERBOSE[1133742896]: Sending to 192.168.10.19 : 5061 
(non-NAT)
Jul  5 09:08:29 VERBOSE[1133742896]: Found RTP audio format 4
Jul  5 09:08:29 VERBOSE[1133742896]: Found RTP audio format 0
Jul  5 09:08:29 VERBOSE[1133742896]: Found RTP audio format 8
Jul  5 09:08:29 VERBOSE[1133742896]: Found RTP audio format 101
Jul  5 09:08:29 VERBOSE[1133742896]: Found video format UNKN
Jul  5 09:08:29 VERBOSE[1133742896]: Found video format UNKN
Jul  5 09:08:29 VERBOSE[1133742896]: Peer RTP is at port 192.168.10.19:9840
Jul  5 09:08:29 VERBOSE[1133742896]: Found description format 
telephone-events
Jul  5 09:08:29 VERBOSE[1133742896]: Found description format H263
Jul  5 09:08:29 VERBOSE[1133742896]: Found description format H261
Jul  5 09:08:29 VERBOSE[1133742896]: Capabilities: us - 0x1(G723), peer 
- audio=0xd(G723|ULAW|ALAW)/video=0xc0000(H261|H263), combined - 0x1(G723)
Jul  5 09:08:29 VERBOSE[1133742896]: Non-codec capabilities: us - 
0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Jul  5 09:08:29 VERBOSE[1133742896]: Found peer '9722352287'
Jul  5 09:08:29 DEBUG[1133742896]: Setting NAT on RTP to 0
Jul  5 09:08:29 DEBUG[1133742896]: Setting NAT on VRTP to 0
Jul  5 09:08:29 DEBUG[1133742896]: Check for res for 9722352287
Jul  5 09:08:29 DEBUG[1133742896]: 9722352287 is not a local user
Jul  5 09:08:29 VERBOSE[1133742896]: Looking for 9722352288 in sip
Jul  5 09:08:30 DEBUG[1133742896]: build_route: Contact hop: 
<sip:9722352287 at 192.168.10.19:5061;user=phone>
Jul  5 09:08:30 VERBOSE[1133742896]: list_route: hop: 
<sip:9722352287 at 192.168.10.19:5061;user=phone>
Jul  5 09:08:30 VERBOSE[1133742896]: Transmitting (no NAT):
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.10.19:5061

From: <sip:9722352287 at sipusers.com;user=phone>;tag=20f8c-176cb1

To: <sip:9722352288 at sipusers.com;user=phone>;tag=as6eeffd24

Call-ID: 50fab160-40794-c0a81dd8 at sipusers.com

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:9722352288 at 192.168.10.158>

Content-Length: 0




 to 192.168.10.19:5061
Jul  5 09:08:30 VERBOSE[1209277232]:     -- Executing 
Dial("SIP/9722352287-32ef", 
"SIP/9722352288|30|r") in new stack
Jul  5 09:08:30 DEBUG[1209277232]: SIMPLE DIAL (NO URL)
Jul  5 09:08:30 DEBUG[1209277232]: Setting NAT on RTP to 0
Jul  5 09:08:30 DEBUG[1209277232]: Setting NAT on VRTP to 0
Jul  5 09:08:30 DEBUG[1209277232]: Outgoing Call for 9722352288
Jul  5 09:08:30 DEBUG[1209277232]: 9722352288 is not a local user
Jul  5 09:08:30 VERBOSE[1209277232]: We're at 192.168.10.158 port 13818
Jul  5 09:08:30 VERBOSE[1209277232]: Video is at 192.168.10.158 port 31668
Jul  5 09:08:30 VERBOSE[1209277232]: Answering/Requesting with root 
capability 1
Jul  5 09:08:30 VERBOSE[1209277232]: Answering with non-codec capability 
0x1(G723)
Jul  5 09:08:30 VERBOSE[1209277232]: 12 headers, 10 lines
Jul  5 09:08:30 VERBOSE[1209277232]: Reliably Transmitting:
INVITE sip:9722352288 at 192.168.10.20:5061 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.158:5060;branch=z9hG4bK339687e8

From: "9722352287" <sip:9722352287 at 192.168.10.158>;tag=as50ef7b7a

To: <sip:9722352288 at 192.168.10.20:5061>

Contact: <sip:9722352287 at 192.168.10.158>

Call-ID: 4df028c320569b3b6c7a62b875bb35f8 at 192.168.10.158

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Date: Mon, 05 Jul 2004 14:08:30 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Content-Type: application/sdp

Content-Length: 218



v=0

o=root 9587 9587 IN IP4 192.168.10.158

s=session

c=IN IP4 192.168.10.158

t=0 0

m=audio 13818 RTP/AVP 4 101

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

 (no NAT) to 192.168.10.20:5061
Jul  5 09:08:30 VERBOSE[1209277232]:     -- Called 9722352288
Jul  5 09:08:30 VERBOSE[1209277232]: Transmitting (no NAT):
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.10.19:5061

From: <sip:9722352287 at sipusers.com;user=phone>;tag=20f8c-176cb1

To: <sip:9722352288 at sipusers.com;user=phone>;tag=as6eeffd24

Call-ID: 50fab160-40794-c0a81dd8 at sipusers.com

CSeq: 101 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:9722352288 at 192.168.10.158>

Content-Length: 0




 to 192.168.10.19:5061
Jul  5 09:08:30 VERBOSE[1133742896]:

Sip read:
REGISTER sip:sipusers.com SIP/2.0

Call-ID: 5cd8-40a18-c0a81dd8 at sipusers.com

From: <sip:9722352287 at sipusers.com;user=phone>;tag=21210-1b31

To: <sip:9722352287 at sipusers.com;user=phone>

CSeq: 222 REGISTER

Via: SIP/2.0/UDP 192.168.10.19:5061

Contact: <sip:9722352287 at 192.168.10.19:5061;user=phone>

Max-Forwards: 70

Expires: 75

Supported: timer

Authorization: Digest 
username="9722352287",realm="asterisk",uri="sip:sipusers.com",response="d062dc737803fcf362f74516a3db6f78",nonce="127dd9f4"

Content-Length: 0
==================================================




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