[pjsip] STUN & local call

Michael Bradley Jr mbradley.jr at gmail.com
Fri Nov 9 09:20:07 EST 2007


Hi Logan,

yes the purpose is a B2B call.
Can you provide the asked hint ;)
Cheers
Mike

logan wrote:
> Hi,
> 
> Your case is not clear to me, are you trying to make B2B SIP calls? Or
> your proxy is local but with STUN being enabled you are having
> problems?
> 
> Thanks.
> 
> Best Regards,
> Hitesh
> 
> On Nov 4, 2007 5:24 PM, Benny Prijono <bennylp at pjsip.org> wrote:
>> Michael Bradley Jr wrote:
>>> Hi,
>>>
>>> when STUN is enable, i can't make a successful call to an UA in the same Network
>>> segment.
>>>
>>> Here is call flow
>>>
>>> UA1: pjsua 0.7.0
>>> localIP: 192.168.0.23 publicIP: 88.xx.xx.xx
>>>
>>> UA2: Grandstream IP-Phone
>>> localIP: 192.168.0.32 publicIP: 88.xx.xx.xx
>>>
>>>
>>> UA1 ---> UA2 (192.168.0.23 >> 192.168.0.32)
>>> INVITE sip:xvc at 192.168.0.32 SIP/2.0
>>> sdp: c:88.xx.xx.xx
>>>
>>> UA2 --> UA1 (192.168.0.32 --> 88.xx.xx.xx)
>>> 200/2.0/UDP 88.xx.xx.xx:5060
>>> sdp: c: 192.168.0.32
>>>
>>>
>>> UA1 never get the 200 since it's send to the public Address...
>>>
>>> So my question:
>>> how can i set the IP of the destination of the RTP stream in my initial
>>> INVITE using pjsua-lib since i can figure out that both UA are behind the same NAT?
>> Currently you can't. When STUN is used, the public IP will always be
>> used in the SDP, regardless of where the destination UA is located.
>> But if only the other UA (UA2) is also pjsua, then there shouldn't
>> be any problem with media communication since pjsua will switch the
>> destination RTP/RTCP address to the source address of the packet.
>>
>> And when you use pjsua to pjsua, you can enable ICE as well. With
>> ICE enabled, it will automatically select which media address to use
>> based on ICE negotiation.
>>
>> cheers,
>>  -benny
>>
>>
>>
>>> Thanks,
>>> Michael
>>
>>
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