|
Hi Guys,
I have following scenario which causes an issue
related to codecs (please look below)
[asterisk] -> Quintum CRSP* / Quintum CMS -> PSTN * Quintum Call Relay SP (CRSP - http://www.quintum.com/main/servproducts.html?id=15), Quintum CMS - H323 based gateway When a call is being placed using a SIP client, UAC (sip client) sends a list of supported codecs to the asterisk and combined list is build as supposed, e.g. Capabilities: us - 0x10d (g723|ulaw|alaw|g729),
peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c
(ulaw|alaw|g729)
My Asterisk has following codec settings
applied:
disallow=all
allow=g723 allow=g729 allow=alaw allow=ulaw Asterisk forwarding calls with prefix 00 to
Quintum SIP-PSTN (Quintum CRSP) gateway which has enabled g723,g729, ulaw
and alaw codecs - the same as ones on the asterisk side.
Quintum CRSP sends back only the first codec out of
the supported codecs list on my side (asterisk) and within the combined
list there is only the first codec on asterisk side. Please
find below log out of described above behaviour:
Sip read:
SIP/2.0 200 OK CSeq: 102 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp From: "pavel"<sip:[EMAIL PROTECTED]:5060>;tag=as74cec9db To: <sip:[EMAIL PROTECTED]>;tag=3ef4af85-1112b Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK5568aa34 Content-Length: 168 User-Agent: Quintum/1.0.0 v=0
o=Quintum 33034 31527 IN IP4 1.2.3.4 s=VoipCall c=IN IP4 1.2.3.4 t=0 0 m=audio 10858 RTP/AVP 4 c=IN IP4 1.2.3.4 a=rtpmap:4 g723/8000/1 10 headers, 8 lines
Found RTP audio format 4 Peer audio RTP is at port 1.2.3.4 :10858 Found description format g723 Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x1 (g723) where us is asterisk, peer is the
Quintum CRSP
This way if the SIP client supports only G729 the
call fails since the combined list of codecs would be again g723 as long as the
g729 is not the first in the list of asterisk's codecs.
Is there any way I can forcably determine the codecs reported for the peer
out of the asterisk's codec capabilities list?
Any idea how I could force the asterisk to
change the order of supported codecs according to the SIP client first codec in
the list or how I could force the Quintum to send back the full list of
supported codecs ?
Thanks and regards,
Pavel |
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