Re: [Asterisk-Users] OH323 incoming audio stutter
Hi Michael and Tony I have the same problem here and I have been able to check that this problem can be solved disabling VAD in h323 destination routers, I think this is a common problem with h323 and oh323 modules users and for me has become a nightmare because my service provider can no longer disable VAD support independently for my connection... I will appreciate if you can include it in some future releases. regards rafael risco Millicom Peru On 4/22/05, Michael Manousos [EMAIL PROTECTED] wrote: Hi Tony, Can you get an ethereal trace of the RTP packets on both directions? A short analysis of those streams (from within the ethereal tools) would help us find the problem. Michael. Tony Mountifield wrote: I'm using asterisk-oh323-0.6.5 with the Janus patch 4 versions of pwlib (v1.6.6.3) and openh323 (v1.13.5.3), and using it to connect to my provider's switch. The effect that I am seeing is that a call starts off fine, but suddenly after a few minutes the audio coming into Asterisk via OH323 gets very broken up to the point of being about 90% silence with occasional brief snippets of audio getting through. When this happens, the audio going out from Asterisk to the other end is still fine, with no disturbances. I have observed this both when using SIP for the local leg of the call and when using IAX. I'm not really sure where to look to diagnose this, not whether it is likely to be an Asterisk problem or something in the switch. Any advice would be appreciated! Cheers Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- rrgv ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 incoming audio stutter
Hi Tony, Can you get an ethereal trace of the RTP packets on both directions? A short analysis of those streams (from within the ethereal tools) would help us find the problem. Michael. Tony Mountifield wrote: I'm using asterisk-oh323-0.6.5 with the Janus patch 4 versions of pwlib (v1.6.6.3) and openh323 (v1.13.5.3), and using it to connect to my provider's switch. The effect that I am seeing is that a call starts off fine, but suddenly after a few minutes the audio coming into Asterisk via OH323 gets very broken up to the point of being about 90% silence with occasional brief snippets of audio getting through. When this happens, the audio going out from Asterisk to the other end is still fine, with no disturbances. I have observed this both when using SIP for the local leg of the call and when using IAX. I'm not really sure where to look to diagnose this, not whether it is likely to be an Asterisk problem or something in the switch. Any advice would be appreciated! Cheers Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OH323 incoming audio stutter
I'm using asterisk-oh323-0.6.5 with the Janus patch 4 versions of pwlib (v1.6.6.3) and openh323 (v1.13.5.3), and using it to connect to my provider's switch. The effect that I am seeing is that a call starts off fine, but suddenly after a few minutes the audio coming into Asterisk via OH323 gets very broken up to the point of being about 90% silence with occasional brief snippets of audio getting through. When this happens, the audio going out from Asterisk to the other end is still fine, with no disturbances. I have observed this both when using SIP for the local leg of the call and when using IAX. I'm not really sure where to look to diagnose this, not whether it is likely to be an Asterisk problem or something in the switch. Any advice would be appreciated! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OH323 incoming audio stutter
The effect that I am seeing is that a call starts off fine, but suddenly after a few minutes the audio coming into Asterisk via OH323 gets very broken up to the point of being about 90% silence with occasional brief snippets of audio getting through. hi, any errors or warnings in Asterisk console? more info please... __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
