Hi,
We identified that ASTPP doesn’t forward the sendrecv property to the customer, while the carrier is sending it.
This is the scenario: the customer, A party, originates a call to any number; ASTPP receives the call, forwards the request to any carrier, the carrier answers with the proper 183 (including the sendrecv), then ASTPP forwards the 183 to the A party but without the sendrecv property in the SDP.
This is affecting the early media and the ability of some autodialers to identify the voicemail (AMD).
This is how the SDP of the carrier looks like:
recv 747 bytes from udp/[carrier.ip.v4.add]:5060 at 09:12:31.815027:
------------------------------------------------------------------------
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP astpp.ip.v4.add;branch=z9hG4bK3j9UKQS0Uc8DN;received=astpp.ip.v4.add;rport=5060
From: "51995992664" <sip:a-number@astpp.ip.v4.add>;tag=0pcFmerQSBF0N
To: <sip:b-number@carrier.ip.v4.add>;tag=sbcsipuas_1_C46184_20260917101230631_b02sbc02
Call-ID: a94574ed-2d44-1240-2798-02da8e1f18c9
Contact: <sip:carrier.ip.v4.add:5060>
CSeq: 120273383 INVITE
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,INFO,UPDATE
Server: sbc_138
Content-Type: application/sdp
Content-Length: 226
v=0
o=SBCSIPUAS 1047122486 1 IN IP4 carrier.ip.v4.add
s=SBCSIPUAS SIP STACK v1.0
c=IN IP4 carrier.ip.v4.add
t=0 0
m=audio 21862 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv <<<<<------HERE IS THE SDP PROPERTY FROM THE CARRIER
-----------------------------------------------------------------------
This is how the SDP from ASTPP to the customer looks like:
send 1001 bytes to udp/[a.party.ip.add]:5060 at 09:12:32.201124:
------------------------------------------------------------------------
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP a.party.ip.add:5060;branch=z9hG4bK7e954ca7;rport=5060
From: <sip:a-number@a.party.ip.add>;tag=as41f9f2dc
To: <sip:b-number@astpp.ip.v4.fqdn:5060>;tag=y4SXgrpgZS2te
Call-ID: 5c7cfd1748932a1b7f83e30b1995141a@a.party.ip.add:5060
CSeq: 103 INVITE
Contact: <sip:51951183372@astpp.ip.v4.add:5060;transport=udp>
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 210
v=0
o=FreeSWITCH 1619739356 1619739358 IN IP4 astpp.ip.v4.add
s=FreeSWITCH
c=IN IP4 astpp.ip.v4.add
t=0 0
m=audio 17190 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<<<<<------HERE IS WEHRE ASTPP MISS THE SDP PROPERTY 'senrecv' TO THE CUSTOMER
------------------------------------------------------------------------
I have tried to edit /opt/ASTPP/freeswitch/scripts/astpp/scripts/astpp.xml.lua with several different options like:
line 56:
...
table.insert(xml, [[<action application="set" data="continue_on_fail=TRUE"/>]]);
-- FORCE EARLY MEDIA & SDP DIRECTION ATTRIBUTES HERE --
--table.insert(xml, [[<action application="set" data="proxy_media=true"/>]]);
--table.insert(xml, [[<action application="set" data="ignore_early_media=false"/>]]);
--table.insert(xml, [[<action application="pre_answer"/>]]);
--table.insert(xml, [[<action application="export" data="sip_append_audio_sdp=\na=sendrecv"/>]]);
--table.insert(xml, [[<action application="set" data="sip_provisional_media_attributes=sendrecv"/>]]);
--table.insert(xml, [[<action application="set" data="media_mix_inbound_outbound_sdp=true"/>]]);
table.insert(xml, [[<action application="set" data="account_id=]]..customer_userinfo['id']..[["/>]]);
table.insert(xml, [[<action application="set" data="parent_id=]]..customer_userinfo['reseller_id']..[["/>]]);
table.insert(xml, [[<action application="set" data="entity_id=]]..customer_userinfo['type']..[["/>]]);
table.insert(xml, [[<action application="set" data="call_processed=internal"/>]]);
table.insert(xml, [[<action application="set" data="call_direction=]]..call_direction..[["/>]]);
...
The commented lines test different combinations of actions, but none worked; the SDP still lacks the sendrcv option.
Also, we have tried adding settings to the sofia profile through the GUI:
media-option=media_direction_in_manifest,resume-media-on-hold
inbound-bypass-media=false
inbound-proxy-media=true
inbound-late-negotiation=true
I’m not sure what else to try; any help or ideas would be welcome. Thank you!
David
Where to Look in an ASTPP Setup
What to Look For