SIP IP Phone to Analog Phone via Gateway

Call Trace

The following trace shows a call originating from an on-network SIP phone

 

and being routed through a gateway to the PSTN.

-----------------------------------------------------------------

SIP Headers

-----------------------------------------------------------------

sip-req:

INVITE sip:93831073@192.168.36.180 SIP/2.0

[192.168.6.20:50753-

>192.168.36.180:5060]

 

Header:

Via: SIP/2.0/UDP 192.168.6.20:5060

 

Header:

From: sip:5120@192.168.6.20

 

Header:

To: <sip:93831073@192.168.36.180>

 

Header:

Call-ID: c2943000-23e062-2e278-2e323931@192.168.6.20

Header:

CSeq: 100 INVITE

 

Header:

Expires: 180

 

Header:

User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabled

Header:

Accept: application/sdp

 

Header:

Contact: sip:5120@192.168.6.20:5060

Header:

Content-Type: application/sdp

 

Header:

Content-Length: 218

 

-----------------------------------------------------------------

 

SDP Headers

 

-----------------------------------------------------------------

Header:

v=0

 

Header:

o=CiscoSystemsSIP-IPPhone-UserAgent 21012 9466 IN IP4 192.168.6.20

Header:

s=SIP Call

 

Header:

c=IN IP4 192.168.6.20

 

Header:

t=0 0

 

Header:

m=audio 25776 RTP/AVP 0 101

 

Header:

a=rtpmap:0 pcmu/8000

 

Header:

a=rtpmap:101 telephone-event/8000

 

Header:

a=fmtp:101 0-11

 

-----------------------------------------------------------------

 

SIP Headers

 

-----------------------------------------------------------------

sip-res:

SIP/2.0 100 Trying [192.168.36.180:5060->192.168.6.20:5060]

Header:

Via: SIP/2.0/UDP 192.168.6.20:5060

 

Header:

From: <sip:5120@192.168.6.20:5060>

 

Header:

To: <sip:93831073@192.168.36.180:5060>

Header:

Call-ID: c2943000-23e062-2e278-2e323931@192.168.6.20

Header:

CSeq: 100 INVITE

 

Header:

Content-Length: 0

 

-----------------------------------------------------------------

 

SIP Headers

 

-----------------------------------------------------------------

sip-req:

INVITE sip:93831073@192.168.36.200:5060;user=phone SIP/2.0

[192.168.36.180:5060->192.168.36.200:5060]

 

Header:

Via: SIP/2.0/UDP 192.168.36.180:5060;branch=1

Header:

Via: SIP/2.0/UDP 192.168.6.20:5060

 

Header:

From: <sip:5120@192.168.6.20:5060>

 

Header:

To: <sip:93831073@192.168.36.180:5060>

Header:

Call-ID: c2943000-23e062-2e278-2e323931@192.168.6.20

Header:

CSeq: 100 INVITE

 

Header:

Proxy-Authorization: Basic VovidaClassXSwitch

Header:

Expires: 180

 

Header:

Record-Route:

 

<sip:93831073@192.168.36.180:5060;maddr=192.168.36.180>

 

Header:

Contact: <sip:5120@192.168.6.20:5060>

Header:

Content-Type: application/sdp

 

Header:

Content-Length: 218

 

-----------------------------------------------------------------

 

SDP Headers

 

-----------------------------------------------------------------

Header:

v=0

 

Header:

o=CiscoSystemsSIP-IPPhone-UserAgent 21012 9466 IN IP4 192.168.6.20

Header:

s=SIP Call

 

Header:

c=IN IP4 192.168.6.20

 

Header:

t=0 0

 

Header:

m=audio 25776 RTP/AVP 0 101

 

Header:

a=rtpmap:0 pcmu/8000

 

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Cisco Systems 1.3.0 manual Being routed through a gateway to the Pstn