Kasun Daminda
2009-Oct-19 13:55 UTC
[asterisk-users] VoIP interconnection with Acme packet SBC
Dear all,
I have found a issue when connecting my asterisk soft switch with Acme
packet SBC.
1) No problem for outgoing calls. ie asterisk to Acme SBC
2) Problem is at incoming. ie Acme to Asterisk
3) My asterisk is connected to a PSTN switch via SS7 with digium interface.
4) When I getting a call from Acme, call is connected, but once answered it
is disconnected.
5) I have taken lot of traces and found that session version that I sends on
SDP/session progress is not similar to session version on SDP/200 OK
message.
Acme also complain me, according to RFC3261, the SDP header should be
identical for both 200 OK and Session progress.
See bullet #2 in ch 13.2.1 (RFC 3261):
o If the initial offer is in an INVITE, the answer MUST be in a
reliable non-failure message from UAS back to UAC which is
correlated to that INVITE. For this specification, that is
only the final 2xx response to that INVITE. *That same exact*
* answer *MAY also be placed in any provisional responses sent
prior to the answer. The UAC MUST treat the first session
description it receives as the answer, and MUST ignore any
session descriptions in subsequent responses to the initial
INVITE.
6) This was quote from the traces
*SDP at Session Progress (183 Session Progress, with session description)*
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 26823 26823 IN IP4
203.189.191.138
Owner Username: root
Session ID: 26823
Session Version: *26823*
*SDP at 200OK*
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 26823 26824 IN IP4
203.189.191.138
Owner Username: root
Session ID: 26823
Session Version: *26824*
**
Hope this info will enough to understand my question. I am grateful to you
if you can help me on this to solve the issue very soon.
**
*Kind Rgds*
*Daminda*
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Vijay Gandhi
2009-Oct-19 14:19 UTC
[asterisk-users] VoIP interconnection with Acme packet SBC
Recheck on the Codec Acme is sending you and you have allowed on your
asterisk box, issue might be codec mismatch.
Regards
Vijay Gandhi
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kasun Daminda
Sent: Monday, October 19, 2009 7:26 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] VoIP interconnection with Acme packet SBC
Dear all,
I have found a issue when connecting my asterisk soft switch with Acme
packet SBC.
1) No problem for outgoing calls. ie asterisk to Acme SBC
2) Problem is at incoming. ie Acme to Asterisk
3) My asterisk is connected to a PSTN switch via SS7 with digium interface.
4) When I getting a call from Acme, call is connected, but once answered it
is disconnected.
5) I have taken lot of traces and found that session version that I sends on
SDP/session progress is not similar to session version on SDP/200 OK
message.
Acme also complain me, according to RFC3261, the SDP header should be
identical for both 200 OK and Session progress.
See bullet #2 in ch 13.2.1 (RFC 3261):
o If the initial offer is in an INVITE, the answer MUST be in a
reliable non-failure message from UAS back to UAC which is
correlated to that INVITE. For this specification, that is
only the final 2xx response to that INVITE. That same exact
answer MAY also be placed in any provisional responses sent
prior to the answer. The UAC MUST treat the first session
description it receives as the answer, and MUST ignore any
session descriptions in subsequent responses to the initial
INVITE.
6) This was quote from the traces
SDP at Session Progress (183 Session Progress, with session description)
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 26823 26823 IN IP4
203.189.191.138
Owner Username: root
Session ID: 26823
Session Version: 26823
SDP at 200OK
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 26823 26824 IN IP4
203.189.191.138
Owner Username: root
Session ID: 26823
Session Version: 26824
Hope this info will enough to understand my question. I am grateful to you
if you can help me on this to solve the issue very soon.
Kind Rgds
Daminda
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Kasun Daminda
2009-Oct-22 07:20 UTC
[asterisk-users] VoIP interconnection with Acme packet SBC
Dear all,
I fixed the issue by myself.
I have edited chan_sip.c file to avoid sdp version gettng increment.
I think this is a bug of asterisk. According to RFCs it should increment it
only it there is change on SDP message body. chan_sip.c alway increase it by
one at every SDP message. I have edited the below part
/* Set RTP Session ID and version */
if (!p->sessionid) {
p->sessionid = getpid();
p->sessionversion = p->sessionid;
} else
p->sessionversion*++*;
As......
/* Set RTP Session ID and version */
if (!p->sessionid) {
p->sessionid = getpid();
p->sessionversion = p->sessionid;
} else
p->sessionversion;
I have removed ++. I am not good programmer. But asterisk lover.
I dont know this is the best solution. However I can receive calls from Acme
packet.
And other important thing to tell is THIS IS NOT A CODEC ISSUE.
thanks everybody
kind Rgds
Daminda
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