[Freeswitch-users] sdp header rewrite

[Gabriel Kuri]

Sorry, I should've included that in the original mesg...

The logs I posted were from trunk-r10055. I updated this morning to
trunk-r10081, still the same issue.

~Gabe

Brian West wrote:
> Which version of FreeSWITCH are you using?
> 
> /b
> 
> On Oct 20, 2008, at 2:25 AM, Gabriel Kuri wrote:
> 
>> I'm having an issue with the linksys spa devices when enabling inbound
>> proxy media mode (inbound-proxy-media=true) and late negotiation
>> (inbound-late-negotiation=true) in the sofia profile. The spa
>> immediately sends a BYE when the call is answered by the called party.
>> For whatever reason, it works fine between two linksys devices  
>> directly
>> connected to FS, but when the call goes out to the PSTN via the SIP
>> provider, the spa isn't happy and sends a BYE.
>>
>> After comparing the raw SIP packets on the wire (tcpdump) and between
>> enabling/disabling proxy-media mode and late negotiation, the only
>> difference I notice is the port in the m= line of the SDP header.
>>
>> According to the freeswitch log, the rtp port would be rewritten to
>> 28044 in the sdp header of the SIP packet sent to the spa device.  
>> But on
>> the wire, the port is rewritten to 0, which I'm guessing is why the  
>> spa
>> isn't happy and sending a BYE.
>>
>> Here's the excerpt from the freeswitch log showing FS rewriting the  
>> port
>> to 28044 for the packet going to the spa device.
>>
>>
>> [DEBUG] sofia_glue.c:1003 sofia_glue_tech_patch_sdp()
>> sofia/internal/<phone_number_removed>@mydomain.net Patched SDP
>> ---
>> v=0
>> o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX
>> s=sip call
>> c=IN IP4 XX.XX.XX.XX
>> t=0 0
>> m=audio 24174 RTP/AVP 18 101
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=ptime:20
>>
>> +++
>> v=0
>> o=01Nextone 3587 27824 IN IP4 XX.XX.XX.XX
>> s=sip call
>> c=IN IP4 YY.YY.YY.YY
>> t=0 0
>> m=audio 28044 RTP/AVP 18 101
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=ptime:20
>>
>>
>> However, the packet on the wire reveals FS rewriting the port to  
>> 0  ...
>>
>> v=0.
>> o=01Nextone 7852943629956191733 8120394851828294756 IN IP4  
>> YY.YY.YY.YY.
>> s=sip call.
>> c=IN IP4 YY.YY.YY.YY.
>> t=0 0.
>> m=audio 0 RTP/AVP 96 101.
>> a=rtpmap:96 G729/8000.
>> a=fmtp:96 annexb=no.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>>
>> Is this a bug or is there some other problem?
>>
>> Thanks for the help,
>> Gabe
>>
>>
>>
>>
>>
>>
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> 
> 
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