[Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 152
Wei Su
wsu at leadtek.com
Tue Mar 22 12:35:12 MST 2005
I understand Asterisk is more like a B2BUA. But when this INFO request is
sent to asterisk, asterisk is supposed to bridge the request to the other
endpoint, right? In what situation, it decides to send a reply; in what
situation, it decides to bridge the request?
What is the role of gateway in SIP world, a proxy, a B2BUA or something
else?
Thank you,
Wei
Date: Fri, 18 Mar 2005 12:51:28 -0600
From: Eric Wieling <eric at fnords.org>
Subject: Re: [Asterisk-Users] Asterisk handling of SIP info
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <423B2330.9090705 at fnords.org>
Content-Type: text/plain; charset=us-ascii; format=flowed
Asterisk is not a SIP proxy.
Wei Su wrote:
> We encouter a situation where we need to use SIP info to convey infomation
> for one end point to another endpoint. I use asterisk to do the test and
> find asterisk does not forward the SIP info to another endpoint, but act
as
> UAS and returns a 4xx error message. I think asterisk is not right to
handle
> this SIP info message.
>
> In RFC 3261 Page 70 "This protocol is designed to be extended. Future
> extensions may define new methods and header fields at any time. An
element
> MUST NOT refuse to proxy a request becasue it contains a method or header
> field it does not know about". In this case, asterisk does not understand
> this INFO message, so it acts as a UAS instead of proxy.
>
> How to let asterisk just forward this request to the other endpoint and
> instead processing it as a UAS?
>
> Thank you,
>
> Wei
>
>
>
>
> Here is the log from the asterisk server:
>
> Mar 17 12:01:31 WARNING[2804]: chan_sip.c:6134 receive_info: Unable to
parse
> INFO message
>
>
> Here is the trace:
>
>
> Frame 96 (808 bytes on wire, 808 bytes captured)
> Session Initiation Protocol
> Request-Line: INFO sip:6002 at 192.168.10.90 SIP/2.0
> Method: INFO
> Resent Packet: False
> Message Header
> Call-ID: 60b8596c-4135c-c0a81e68 at 192.168.10.90
> From: Demo2<sip:6003 at 192.168.10.90;user=phone>;tag=221a0-a1cf
> SIP Display info: Demo2
> SIP from address: sip:6003 at 192.168.10.90
> SIP tag: 221a0-a1cf
> To: <sip:6002 at 192.168.10.90;user=phone>;tag=as6b294484
> SIP to address: sip:6002 at 192.168.10.90
> SIP tag: as6b294484
> CSeq: 102 INFO
> Via: SIP/2.0/UDP 192.168.10.164:5060
> Contact: Demo2<sip:6003 at 192.168.10.164:5060;user=phone>
> Max-Forwards: 70
> Supported: timer
> Proxy-Authorization: Digest
>
username="6003",realm="asterisk",uri="sip:6002 at 192.168.10.90",response="034d
> 6b15ec1b2fa91f59c55d51c0a8e7",nonce="70c7fe86"
> Content-Type: application/media_control+xml
> Content-Length: 195
> Message body
> <?xml version="1.0" encoding="utf-8" ?>\n
> <media_control>\n
> <vc_primitive>\n
> <to_encoder>\n
> <picture_fast_update>\n
> </picture_fast_update>\n
> </to_encoder>\n
> </vc_primitive>\n
> </media_control>
>
>
> Frame 97 (430 bytes on wire, 430 bytes captured)
> Session Initiation Protocol
> Status-Line: SIP/2.0 415 Unsupported media type
> Status-Code: 415
> Resent Packet: False
> Message Header
> Via: SIP/2.0/UDP 192.168.10.164:5060
> From: Demo2<sip:6003 at 192.168.10.90;user=phone>;tag=221a0-a1cf
> SIP Display info: Demo2
> SIP from address: sip:6003 at 192.168.10.90
> SIP tag: 221a0-a1cf
> To: <sip:6002 at 192.168.10.90;user=phone>;tag=as6b294484
> SIP to address: sip:6002 at 192.168.10.90
> SIP tag: as6b294484
> Call-ID: 60b8596c-4135c-c0a81e68 at 192.168.10.90
> CSeq: 102 INFO
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:6002 at 192.168.10.90>
> Content-Length: 0
>
>
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