1 way audio SIP/H323

I make a call in the following way

H323 Softphone–>Asterisk–>Sip Operator

For this call flow I get one way audio (H323 client doesn’t hear remote side). As debug shows (rtp debug) Asterisk doesn’t receive RTP stream from Sip operator. I see only packets from H323 and to Sip Operator.

For H323 client I use openphone and SJphone. Asterisk uses ooh323 which is in Asterisk-addons. I tried both 1.2 version and 1.4.

As I see from sip debug Sip Operator uses SER.

I tried to use different codecs ( gsm, g711 etc.). As I see from sip debug codecs are supported by remote side.

if I use these schemes everything is ok:

H323 Softphone–>Asterisk–>Sip Phone

Sip Phone–>Asterisk–>Sip Operator

I don’t use firewalls.

Can anybody help ?
I can show any debugs if u need.

Try manipulating the Faststart, H245Tunneling,H245 in Setup Properties.
Though even after manipulating, some routes could still not work.
This is my experience.


Thanks for answer.

Does it really mean that h.323 leg can affect SIP leg? :confused:

Check bindaddr option in your h323.conf. If it set to default ( and local hostname resolves to - audio will not work. You should specify real IP address in bindaddr option.