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Gareth Blades

2013-Sep-17 10:17 UTC

[asterisk-users] RTP not being switched between both SIP endpoints

We have a system where calls are coming in from telcos via an opensips 
server and then being redirected out to a regular sip destination.
There is no NAT, DTMF features, call recording, or codec translation 
being performed so I would expect asterisk to issue a reinvite after the 
call is answered and switch the audio however it is not happening.

Here is the sip peer information for the call coming from opensips. 
Directmedia is not specifically defined so its using the asterisk 
default value.

   * Name       : vmpubopensips3
   Description  :
   Secret       : <Not set>
   MD5Secret    : <Not set>
   Remote Secret: <Not set>
   Context      : from-pubopensips
   Record On feature : automon
   Record Off feature : automon
   Subscr.Cont. : <Not set>
   Language     :
   Tonezone     : <Not set>
   AMA flags    : Unknown
   Transfer mode: open
   CallingPres  : Presentation Allowed, Not Screened
   Callgroup    :
   Pickupgroup  :
   Named Callgr :
   Nam. Pickupgr:
   MOH Suggest  :
   Mailbox      :
   VM Extension : asterisk
   LastMsgsSent : 0/0
   Call limit   : 0
   Max forwards : 0
   Dynamic      : No
   Callerid     : "" <>
   MaxCallBR    : 384 kbps
   Expire       : -1
   Insecure     : no
   Force rport  : Auto (No)
   Symmetric RTP: No
   ACL          : No
   DirectMedACL : No
   T.38 support : No
   T.38 EC mode : Unknown
   T.38 MaxDtgrm: -1
   DirectMedia  : Yes
   PromiscRedir : No
   User=Phone   : No
   Video Support: No
   Text Support : No
   Ign SDP ver  : No
   Trust RPID   : Yes
   Send RPID    : No
   Subscriptions: Yes
   Overlap dial : No
   DTMFmode     : rfc2833
   Timer T1     : 500
   Timer B      : 32000
   ToHost       : 88.x.x.x
   Addr->IP     : 88.x.x.x:5060
   Defaddr->IP  : (null)
   Prim.Transp. : UDP
   Allowed.Trsp : UDP
   Def. Username:
   SIP Options  : (none)
   Codecs       : (gsm|ulaw|alaw)
   Codec Order  : (alaw:20,ulaw:20,gsm:20)
   Auto-Framing :  No
   Status       : Unmonitored
   Useragent    :
   Reg. Contact :
   Qualify Freq : 60000 ms
   Keepalive    : 0 ms
   Sess-Timers  : Accept
   Sess-Refresh : uas
   Sess-Expires : 1800 secs
   Min-Sess     : 90 secs
   RTP Engine   : asterisk
   Parkinglot   :
   Use Reason   : No
   Encryption   : No

When the call comes in the SDP contains :-

v=0.
o=root 973184584 973184584 IN IP4 81.x.x.x
s=session.
c=IN IP4 81.x.x.x
t=0 0.
m=audio 11370 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

and we reply back with :-

v=0.
o=root 822402971 822402971 IN IP4 88.x.x.x
s=Asterisk PBX 11.2-cert2.
c=IN IP4 88.x.x.x
t=0 0.
m=audio 10428 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


When we send the outbound SIP information we advertise the following SDP :-

v=0.
o=root 431105643 431105643 IN IP4 88.x.x.x
s=Asterisk PBX 11.2-cert2.
c=IN IP4 88.x.x.x
t=0 0.
m=audio 10144 RTP/AVP 8 3 0 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

and the other end replies with :-

v=0.
o=hksbc1a 609621538 609621538 IN IP4 203.x.x.x
s=sip call.
c=IN IP4 203.x.x.x
t=0 0.
m=audio 34146 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=ptime:20.
a=fmtp:101 0-15.

In the Dial() command the only option we are using is M() which is used 
to run a macro when the call is answered. This is used to update cdr 
records and perform other features if they are enabled. In this case we 
are just updating the cdr records so I would expect the audio to be 
switched as soon as the macro finishes.

Any ideas what could be wrong?
We are running Asterisk PBX 11.2-cert2

Thanks
Gareth

Kenny Watson

2013-Sep-18 11:40 UTC

head link

[asterisk-users] RTP not being switched between both SIP endpoints

Hi,

Since opensips is not handling media (i presume) is the audio not already going
direct from asterisk to the other endpoint?

Thanks

Kenny
________________________________________
From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at
lists.digium.com] on behalf of Gareth Blades [mailinglist+asterisk at
dns99.co.uk]
Sent: 17 September 2013 11:17
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] RTP not being switched between both SIP endpoints

We have a system where calls are coming in from telcos via an opensips
server and then being redirected out to a regular sip destination.
There is no NAT, DTMF features, call recording, or codec translation
being performed so I would expect asterisk to issue a reinvite after the
call is answered and switch the audio however it is not happening.

Here is the sip peer information for the call coming from opensips.
Directmedia is not specifically defined so its using the asterisk
default value.

   * Name       : vmpubopensips3
   Description  :
   Secret       : <Not set>
   MD5Secret    : <Not set>
   Remote Secret: <Not set>
   Context      : from-pubopensips
   Record On feature : automon
   Record Off feature : automon
   Subscr.Cont. : <Not set>
   Language     :
   Tonezone     : <Not set>
   AMA flags    : Unknown
   Transfer mode: open
   CallingPres  : Presentation Allowed, Not Screened
   Callgroup    :
   Pickupgroup  :
   Named Callgr :
   Nam. Pickupgr:
   MOH Suggest  :
   Mailbox      :
   VM Extension : asterisk
   LastMsgsSent : 0/0
   Call limit   : 0
   Max forwards : 0
   Dynamic      : No
   Callerid     : "" <>
   MaxCallBR    : 384 kbps
   Expire       : -1
   Insecure     : no
   Force rport  : Auto (No)
   Symmetric RTP: No
   ACL          : No
   DirectMedACL : No
   T.38 support : No
   T.38 EC mode : Unknown
   T.38 MaxDtgrm: -1
   DirectMedia  : Yes
   PromiscRedir : No
   User=Phone   : No
   Video Support: No
   Text Support : No
   Ign SDP ver  : No
   Trust RPID   : Yes
   Send RPID    : No
   Subscriptions: Yes
   Overlap dial : No
   DTMFmode     : rfc2833
   Timer T1     : 500
   Timer B      : 32000
   ToHost       : 88.x.x.x
   Addr->IP     : 88.x.x.x:5060
   Defaddr->IP  : (null)
   Prim.Transp. : UDP
   Allowed.Trsp : UDP
   Def. Username:
   SIP Options  : (none)
   Codecs       : (gsm|ulaw|alaw)
   Codec Order  : (alaw:20,ulaw:20,gsm:20)
   Auto-Framing :  No
   Status       : Unmonitored
   Useragent    :
   Reg. Contact :
   Qualify Freq : 60000 ms
   Keepalive    : 0 ms
   Sess-Timers  : Accept
   Sess-Refresh : uas
   Sess-Expires : 1800 secs
   Min-Sess     : 90 secs
   RTP Engine   : asterisk
   Parkinglot   :
   Use Reason   : No
   Encryption   : No

When the call comes in the SDP contains :-

v=0.
o=root 973184584 973184584 IN IP4 81.x.x.x
s=session.
c=IN IP4 81.x.x.x
t=0 0.
m=audio 11370 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

and we reply back with :-

v=0.
o=root 822402971 822402971 IN IP4 88.x.x.x
s=Asterisk PBX 11.2-cert2.
c=IN IP4 88.x.x.x
t=0 0.
m=audio 10428 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.


When we send the outbound SIP information we advertise the following SDP :-

v=0.
o=root 431105643 431105643 IN IP4 88.x.x.x
s=Asterisk PBX 11.2-cert2.
c=IN IP4 88.x.x.x
t=0 0.
m=audio 10144 RTP/AVP 8 3 0 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.

and the other end replies with :-

v=0.
o=hksbc1a 609621538 609621538 IN IP4 203.x.x.x
s=sip call.
c=IN IP4 203.x.x.x
t=0 0.
m=audio 34146 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=ptime:20.
a=fmtp:101 0-15.

In the Dial() command the only option we are using is M() which is used
to run a macro when the call is answered. This is used to update cdr
records and perform other features if they are enabled. In this case we
are just updating the cdr records so I would expect the audio to be
switched as soon as the macro finishes.

Any ideas what could be wrong?
We are running Asterisk PBX 11.2-cert2

Thanks
Gareth

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