[Asterisk-Users] Sipura line 1 outgoing voice problem?
Mark Musone
mmusone at shatterit.com
Mon Apr 19 19:22:56 MST 2004
I did some packet sniffs, and below are two sets of packets, the first
is the second phone line that works fine with an incoming call and
outgoing sound This seems to be the key packet that sets up the codes
and sessions
( I really don't know any of this sip stuff well, but hopefully somebody
on the list knows it):
The main thing to point out is the initial "Media Description" section.
In the Working line2, it's:
Media Description, name and address (m): audio 16446 RTP/AVP 0 101
...
Media Format: 101
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute Fieldname: fmtp
Media Attribute Value: 101 0-15
So I believe this is setting up the audio transmit stuff. On the line1,
this data is NOT being sent. I don't know what this stuff means, still
looking into it..but maybe someone here does know it? Am I possibly even
on the right track??
The other thought I have, since this is data being sent FROM the Sipura
TO asterisk, the problem is once again seeming to point directly at
Sipura, and it's basically not sending the audio info..
Does any of this even make any sense??
Hope this either helps others to possibly find a fix, or if anyone
_does_ have a fix, please let me know!
Packet for line2, working outgoing audio
Frame 7 (733 bytes on wire, 733 bytes captured)
Ethernet II, Src: 00:0e:08:aa:b7:b1, Dst: 00:07:95:55:7b:ce
Internet Protocol, Src Addr: 192.168.1.21 (192.168.1.21), Dst Addr:
192.168.1.20 (192.168.1.20)
User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060)
Session Initiation Protocol
Status line: SIP/2.0 200 OK
Status-Code: 200
Message Header
To: <sip:2202 at 192.168.1.21:5061>;tag=6b4e39bb53bc50bc
SIP to address: <sip:2202 at 192.168.1.21:5061>
SIP tag: 6b4e39bb53bc50bc
From: "asterisk" <sip:asterisk at 192.168.1.20>;tag=as55a02558
SIP from address: "asterisk" <sip:asterisk at 192.168.1.20>
SIP tag: as55a02558
Call-ID: 5cb768a21438ef471be268b83f183d38 at 192.168.1.20
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK0d8c5d0f
Contact: SPA 2202 <sip:2202 at 192.168.1.21:5061>
Server: Sipura/SPA2000-2.0.2
Content-Length: 210
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 25669620 25669620 IN IP4
192.168.1.21
Owner Username: -
Session ID: 25669620
Session Version: 25669620
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 192.168.1.21
Session Name (s): -
Connection Information (c): IN IP4 192.168.1.21
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 192.168.1.21
Time Description, active time (t): 0 0
Session Start Time: 0
Session Start Time: 0
Media Description, name and address (m): audio 16446 RTP/AVP
0 101
Media Type: audio
Media Port: 16446
Media Proto: RTP/AVP
Media Format: 0
Media Format: 101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute Fieldname: fmtp
Media Attribute Value: 101 0-15
Media Attribute (a): ptime:20
Media Attribute Fieldname: ptime
Media Attribute Value: 20
Media Attribute (a): sendrecv
Below is an incoming phone call to line1, with the outgoing voice NOT
working:
Frame 7 (672 bytes on wire, 672 bytes captured)
Ethernet II, Src: 00:0e:08:aa:b7:b1, Dst: 00:07:95:55:7b:ce
Internet Protocol, Src Addr: 192.168.1.21 (192.168.1.21), Dst Addr:
192.168.1.20 (192.168.1.20)
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)
Session Initiation Protocol
Status line: SIP/2.0 200 OK
Status-Code: 200
Message Header
To: <sip:2201 at 192.168.1.21>;tag=f03d01bbf25c28bb
SIP to address: <sip:2201 at 192.168.1.21>
SIP tag: f03d01bbf25c28bb
From: "asterisk" <sip:asterisk at 192.168.1.20>;tag=as5c261e75
SIP from address: "asterisk" <sip:asterisk at 192.168.1.20>
SIP tag: as5c261e75
Call-ID: 0c7283854ade41761638e7e44d662e98 at 192.168.1.20
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK7b7c931a
Contact: Ext 2201 <sip:2201 at 192.168.1.21:5060>
Server: Sipura/SPA2000-2.0.2
Content-Length: 154
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 25645618 25645618 IN IP4
192.168.1.21
Owner Username: -
Session ID: 25645618
Session Version: 25645618
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 192.168.1.21
Session Name (s): -
Connection Information (c): IN IP4 192.168.1.21
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 192.168.1.21
Time Description, active time (t): 0 0
Session Start Time: 0
Session Start Time: 0
Media Description, name and address (m): audio 16440 RTP/AVP
0
Media Type: audio
Media Port: 16440
Media Proto: RTP/AVP
Media Format: 0
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Attribute Value: 0 PCMU/8000
Media Attribute (a): ptime:20
Media Attribute Fieldname: ptime
Media Attribute Value: 20
Media Attribute (a): sendrecv
Thanks!
-Mark
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