[Freeswitch-users] "Proxy|Bypass Media" Wrong Payload.

Mon Sep 28 21:08:21 PDT 2009

Hi,

I'm having a strange behavior with the FS when I'm using it with   
"inboud-late-negotiation=true"  and with the both scenarios  "proxy- 
media=true" or "bypass-media=true".  The FS is acting as a pseudo  
proxy (I know that it is not intend for that).

The configuration is similar to this:

[Endpoints]   <= a => [FS1] <= b => [FS2]

Where FS1 is acting as a proxy and registrar. The other one will  
simply handle the calls between endpoints or with a Gateway.

Basically the problem is when I'm trying to call to an AddPac  
Endpoint, then FS2 sends a call to the FS1 and this one do the proxy  
or bypass process. Everything works fine with most of the UA  
(Grandstream/Sipura/Linksys/Xlite/Zoiper/etc) but with the AddPac the  
FS1 is returning a Payload 96 which seams to be wrong. The only  
different (In Comparison to the other UA) that I've seen is the  
"a=ptime:20" in the 200/OK/SDP witch is teorically right. I've seen  
some posts with a similar issue las year: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg01468.html

Another interesting point is when the media is handle by FS1 ("proxy- 
media=false"|"bypass-media=false") everything works fine. Also, what  
is more weird is when the call is generated by the AddPac the SDP  
everything works fine too.

Probably is an AddPac issue, due is the only one failing, however  
since I have not found something wrong in the  SIP capture I'm  
starting to have my doubts...

I'm running Freeswitch 1.0.4pre9.

Here is a small flow of the call:

1) FS2 INVITE TO FS1
2) FS1 INVITE TO AddPac Endpoint
3) AddPac Endpoint responds 200 with apparently correct SDP.
4) FS1 responds 200/OK with an Incorrect Payload

Here are the corresponding SIP packet  trace with the  previous call  
flow:

Packet 1
=======
INVITE sip:9499@[FS1 IP] SIP/2.0
Via: SIP/2.0/UDP [FS2 IP]:5060;branch=z9hG4bK60c74dbf;rport
From: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;tag=as3bd06925
To: <sip:9499@[FS1 IP]>
Contact: <sip:[Source Number]@[FS2 IP]>
Call-ID: 14b07c2014d3a0aa7719efa03d979eed@[FS2 IP]
CSeq: 102 INVITE
User-Agent: Legacy
Max-Forwards: 70
Remote-Party-ID: "[Source Number]" <sip:[Source Number]@[FS2  
IP]>;privacy=off;screen=no
Date: Mon, 28 Sep 2009 15:28:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 16330 16330 IN IP4 [FS2 IP]
s=session
c=IN IP4 [FS2 IP]
t=0 0
m=audio 12558 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Packet 2
=======
INVITE sip:9499@[AddPac IP] SIP/2.0
Via: SIP/2.0/UDP [FS1 IP];rport;branch=z9hG4bKXt9pr4er2H2Np
Max-Forwards: 69
From: "[Source Number]" <sip:[Source Number]@[FS1 IP]>;tag=7ZDB1S0pFpSja
To: <sip:9499@[AddPac IP]>
Call-ID: 765f438b-26e6-122d-3490-51d73cb8d94c
CSeq: 120957528 INVITE
Contact: <sip:mod_sofia@[FS1 IP]:5060>
User-Agent: Proxy 1.1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session- 
description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 228
Remote-Party-ID: <[Source Number]>

v=0
o=root 16330 16330 IN IP4 [FS2 IP]
s=session
c=IN IP4 [FS1 IP]
t=0 0
m=audio 27382 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

Packet 3
=======
SIP/2.0 200 OK
Via: SIP/2.0/UDP [FS1 IP];rport;branch=z9hG4bKXt9pr4er2H2Np
From: "[Source Number]" <sip:[Source Number]@[FS1 IP]>;tag=7ZDB1S0pFpSja
To: <sip:9499@[AddPac IP]>;tag=a34a0003a4
Call-ID: 765f438b-26e6-122d-3490-51d73cb8d94c
CSeq: 120957528 INVITE
Supported: timer, replaces, early-session
User-Agent: AddPac SIP Gateway
Contact: sip:9499@[AddPac IP]
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 235

v=0
o=9499 1254144425 1254144425 IN IP4 [AddPac IP]
s=AddPac Gateway SDP
c=IN IP4 [AddPac IP]
t=1254144425 0
m=audio 23004 RTP/AVP 0 101
a=ptime:20
a=rtpmap:0 PCMU/8000/3
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

Packet 4
=======

SIP/2.0 200 OK
Via: SIP/2.0/UDP [FS2 IP]:5060;branch=z9hG4bK60c74dbf;rport=5060
From: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;tag=as3bd06925
To: <sip:9499@[FS1 IP]>;tag=6pmjZyFKjD3Ze
Call-ID: 14b07c2014d3a0aa7719efa03d979eed@[FS2 IP]
CSeq: 102 INVITE
Contact: <sip:mod_sofia@[FS1 IP]:5060;transport=udp>
User-Agent: Proxy 1.1
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session- 
description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 221

v=0
o=9499 1254144425 1254144425 IN IP4 [AddPac IP]
s=AddPac Gateway SDP
c=IN IP4 [FS1 IP]
t=1254144425 0
m=audio 0 RTP/AVP 96 101
a=rtpmap:96 PCMU/8000/3
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
ACK sip:mod_sofia@[FS1 IP]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP [FS2 IP]:5060;branch=z9hG4bK56269b6a;rport
From: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;tag=as3bd06925
To: <sip:9499@[FS1 IP]>;tag=6pmjZyFKjD3Ze
Contact: <sip:[Source Number]@[FS2 IP]>
Call-ID: 14b07c2014d3a0aa7719efa03d979eed@[FS2 IP]
CSeq: 102 ACK
User-Agent: Legacy
Max-Forwards: 70
Remote-Party-ID: "[Source Number]" <sip:[Source Number]@[FS2  
IP]>;privacy=off;screen=no
Content-Length: 0

I've tried the same test with different codecs, and it is always  
messing with the payload.

Thanks in advance
Regards
M

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