1 way audio SIP/H323

Asterisk Asterisk Support
1

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.

2

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

Regards,
Arman

3

Thanks for answer.

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

4

Check bindaddr option in your h323.conf. If it set to default (0.0.0.0) and local hostname resolves to 127.0.0.1 - audio will not work. You should specify real IP address in bindaddr option.