Calls not terminating properly?

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This may not be related, but using the latest PJSIP from subversion,  
I get something similar. If I make a call, terminate it, then make  
another, I'll end up with two calls and garbled audio. (Like it mixes  
the RTP from both.) However, this only happens if I compile with  
debug (-g), otherwise, things work fine.

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com

On Jun 3, 2008, at 5:51 AM, Jo?o C?sar wrote:

> Hell all
>
> I'm working on a project that uses pjsip, pjmedia and pjsua-api. We  
> have a memory port as source of audio, and we are establishing a  
> sip connection to a SIP compliant third party product. However, on  
> call termination from the other side, i think we are not properly  
> handing the hangup of the call since we are getting this kind of  
> errors:
>
> strm0284c64c RTP recv() error: Connection reset by peer  
> (WSAECONNRESET) [err:130054]
> strm0284c64c RTP recv() error: Connection reset by peer  
> (WSAECONNRESET) [err:130054]
> strm0284c64c RTP recv() error: Connection reset by peer  
> (WSAECONNRESET) [err:130054]
>
> assumptions:
> - we are using UDP transport
> - we are adding the ports with pjsua_conf_add_port
> - making the call with pjsua_call_make_call
> - using pjmedia_snd_port_create_player to create the port
> - using pjmedia_snd_port_connect to connect the ports
>
> we have implemented the following callback functions
> - on_call_media_state
> - on_call_state
> - on_incoming_call
>
> However, on the on_call_state we are only printing some call info.  
> do we need to implement explicit hangup here if connection is no  
> longer active?
>
> Best regards
>
> -- 
> Joao Cesar
> msn: jpcesar at gmail.com
> gtalk: jpcesar at gmail.com
> icq: 13790802
> _______________________________________________
> Visit our blog: http://blog.pjsip.org
>
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