ICE: Media not starting in caller

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Benny Prijono wrote:
> On Wed, Aug 13, 2008 at 11:27 AM, Pedro Gon?alves 
> <pedro.pandre at gmail.com <mailto:pedro.pandre at gmail.com>> wrote:
>
>     Hi!
>
>     I already managed to use ICE to establish a call between a client
>     behind
>     symmetric NAT and a client directly (no NAT) connected to my SIP
>     Server,
>     which proves that ICE is working fine :)
>
>     However, I have a problem when ICE can't establish a connection (ex:
>     Port Restricted Cone NAT to Symmetric NAT).
>     I think I am missing something, because the media (RTP and RTCP)
>     is not
>     starting in caller when the initial answer (200 OK INVITE) is received
>     by the caller.
>
>     As I am using PJSIP at a low level, the process flow is like this:
>     [When starting a call]
>
>     == Caller ==
>     pjmedia_ice_create()
>     pjmedia_transport_media_create()
>     pjmedia_transport_encode_sdp()
>     pjsip_inv_create_uac()
>     pjsip_inv_invite()
>     pjsip_inv_send_msg()
>
>     [INVITE goes to answerer]
>
>     == Answerer ==
>     pjsip_inv_answer()
>     pjmedia_ice_create()
>     pjmedia_transport_media_create()
>     pjmedia_transport_encode_sdp()
>     pjsip_inv_send_msg()
>     pjmedia_transport_media_start()
>
>     [200 OK goes to caller]
>
>     == Caller ==
>     on_media_update():
>       pjmedia_transport_media_start()
>
>     on_ice_complete():
>       pjmedia_endpt_create_sdp()
>       pjmedia_transport_encode_sdp()
>       pjsip_inv_reinvite()
>       pjsip_inv_send_msg()
>     (...)
>
>     With this process flow, I can see that the default candidates (relayed
>     candidates) are chosen and negotiated; I can even see the media flow
>     going from answerer to relay sever to caller, but not the other way
>     around (caller to relay server to answerer).
>     Any idea why?
>
>
> Nope.
>  
>
>     Am I missing something?
>
>
> Yes. The log. :)
Here they are, in attachment.
>
> And btw, in your log, all SIP messages are written as one very long 
> line and this makes it very hard to read. Can you fix that first?
>
Sorry, I can see the log perfectly with Notepad or PSPad.
Here is an example of a sip message:
 15:43:33.640            LOG  RX 354 bytes Response msg 
100/REGISTER/cseq=18510 (rdata0AAD805C) from UDP 172.18.0.170:5060:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 
10.0.0.2:1470;rport=1470;branch=z9hG4bKPj9663ef010ac74fe5bdb59f8a3731d622;received=172.18.0.78

From: 
<sip:+351919414978 at webchamada.vodafone.pt>;tag=47530169257643ef812a6940a8fe5174

To: <sip:+351919414978 at webchamada.vodafone.pt>

Call-ID: da82eb3395394dfc8bc0b163cd1e01ac

CSeq: 18510 REGISTER

Content-Length: 0


Best Regards
Pedro Gon?alves
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