[pjsip] No RTP over UMTS

Benny Prijono bennylp at pjsip.org
Tue Feb 19 04:02:05 EST 2008


On 2/14/08, Tzury Bar Yochay <tzury.by at gmail.com> wrote:
> Topology:
>
> SoftPhone on WinXP @ 10.0.0.201
> Asterisk on VirtualBox on this WinXP (Bridged) @ 10.0.0.211
> HTC SmartPhone @ 192.118.11.112 (STUN is active)
>
> When trying to connect via GPRS, I Can see the following symptoms:
>
>    1. RTP from the SmartPhone (192.118.11.112) is not reaching the asterisk,
>    2. RTP from Desktop SoftPhone to 192.118.11.112 dose not heard in the
> SmartPhone
>
> I would add that when connecting with WiFi (same LAN) everything works fine
> [thanks for the STUN tip Angelos;-) ].
> I attached WireShark dump in case anyone would like to take a look at.

I think the fact that Asterisk is running behind NAT makes it more
difficult for VoIP traffic to get through. Try registering to a public
SIP proxy (say Iptel.org), and call pjsua in the desktop instead of
Asterisk, and enable STUN and ICE on both pjsua and the HTC
SmartPhone. If RTP can't get through in this config, then the NAT
configuration in the middle is just too unfriendly for VoIP. Wait
until we have TURN in pjsip. ;-)

cheers,
 -benny




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