[Freeswitch-users] SDP issue receiving calls from SIP connection
Kirk Bateman
kirk.bateman at gmail.com
Mon Aug 11 08:09:43 PDT 2008
Anthony,
Yes I'd figured that out :)
I couldn't see anything specific that it was complaining about ... I've
looked at the source and not figured it out yet... (really must try and
memorize the spec someday).
I was wondering if it was something to do with the X-NSE bit (dtmf tones
extension to rfc ??) but given that I've set it to late negotiation I
wouldn't expect the SDP parser to complain about that.
I'm hoping the sofia dev can point me in the right direction.
I have since I wrote the original mail managed to test (without any real
changes) that I can make outgoing calls using the console originate command,
that worked (no audio but I expected that).
Cheers
Kirk
2008/8/11 Anthony Minessale <anthony.minessale at gmail.com>
> I assume it's "whining" about the SDP parse error encountered.
> nua(0x8120fe0): INVITE server: error parsing SDP
>
> I do not have the spec memorized but I'm sure if we show it to the sofia
> dev he can tell us the specific problem with the sdp.
>
>
>
>
> On Mon, Aug 11, 2008 at 8:30 AM, Kirk Bateman <kirk.bateman at gmail.com>wrote:
>
>> Afternoon everyone,
>>
>> I have a bit of a problem with Freeswitch receiving RTP INVITEs from my
>> SIP provider (tesco's internet phone ... I know SIP isn't supported
>> technically, but it sort of works...)
>>
>> Freeswitch (sofia) seems to be whinging about the INVITE SDP format ...
>> but I'm not sure why ... here's a dump of my log / trace, anyone got any
>> clues, I've tried adding extra codecs to my list and setting the late
>> negotiation but that doesn't seem to fix it ... (registration all works fine
>> by the way ...)
>>
>>
>> ------------------------------------------------------------------------
>> INVITE sip:MY_PHONE_NUMBER at MY_IP_ADDRESS:5060;transport=udp SIP/2.0
>> Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1
>> Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022
>> CSeq: 100 INVITE
>> From: "ANOTHER_PHONE_NUMBER" <
>> sip:ANOTHER_PHONE_NUMBER at sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER at sip.tescointernetphone.com>
>> ;user=phone>;tag=SDvke8201-b2b.31deb49
>> To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER at sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER at sip.tescointernetphone.com>
>> ;user=phone>
>> Max-Forwards: 69
>> Content-Type: application/sdp
>> Contact: <sip:ANOTHER_PHONE_NUMBER at 77.75.1.10:5060;transport=udp>
>> Allow: INVITE,CANCEL,ACK,BYE
>> Accept: application/sdp
>> Content-Length: 840
>>
>> v=0
>> o=- 3250588766 0 IN IP4 77.75.1.10
>> s=-
>> c=IN IP4 77.75.1.10
>> t=0 0
>> m=audio 54424 RTP/AVP 18 99 101 102 103 15 104 4 105 106 107 108 125
>> 109 100 8 0
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:99 G.729a/8000
>> a=rtpmap:101 G.726-16/8000
>> a=rtpmap:102 G.726-24/8000
>> a=rtpmap:103 G.726-32/8000
>> a=rtpmap:15 G728/8000
>> a=rtpmap:104 G.723.1-H/8000
>> a=rtpmap:4 G723/8000
>> a=rtpmap:105 G.723.1-L/8000
>> a=rtpmap:106 G.729b/8000
>> a=rtpmap:107 G.723.1a-H/8000
>> a=rtpmap:108 G.723.1a-L/8000
>> a=rtpmap:125 G.nX64/8000
>> a=rtpmap:109 AMR/8000
>> a=rtpmap:100 X-NSE/8000
>> a=fmtp:100 192-194,200-202
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> a=maxptime:20
>> a=maxptime:30
>> a=ptime:20
>> a=ptime:30
>> a=X-cap: 1 audio RTP/AVP 100
>> a=X-cap: 2 image udptl t38
>> a=X-sqn:0
>> a=X-cpar: a=rtpmap:100 X-NSE/8000
>> a=X-cpar: a=fmtp:100 192-194,200-202
>>
>> ------------------------------------------------------------------------
>> tport_deliver(0x80fcbf0): msg 0x8129468 (1416 bytes) from udp/
>> 77.75.1.10:5060/sip next=(nil)
>> nta: received INVITE sip:MY_PHONE_NUMBER at MY_IP_ADDRESS:5060;transport=udp
>> SIP/2.0 (CSeq 100)
>> nta: canonizing sip:MY_PHONE_NUMBER at MY_IP_ADDRESS:5060 with contact
>> nta: INVITE (100) going to a default leg
>> nta: timer set to 200 ms
>> nua: nua_stack_process_request: entering
>> nua: nh_create: entering
>> nua: nh_create_handle: entering
>> nua: nua_stack_set_params: entering
>> soa_clone(static::0x80ef908, 0x80eccb8, 0x8120fe0) called
>> soa_set_params(static::0x8111d08, ...) called
>> nta_leg_tcreate(0x81200c0)
>> soa_init_offer_answer(static::0x8111d08) called
>> soa_set_remote_sdp(static::0x8111d08, (nil), 0x810ed20, 840) called
>> nua(0x8120fe0): INVITE server: error parsing SDP
>> nua: nua_invite_server_respond: entering
>> tport_tsend(0x80fcbf0) tpn = UDP/77.75.1.10:5060
>> tport_resolve addrinfo = 77.75.1.10:5060
>> tport_by_addrinfo(0x80fcbf0): not found by name UDP/77.75.1.10:5060
>> tport_vsend returned 661
>> send 661 bytes to udp/[77.75.1.10]:5060 at 13:16:28.358779:
>>
>> ------------------------------------------------------------------------
>> SIP/2.0 400 Bad Session Description
>> Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1
>> From: "ANOTHER_PHONE_NUMBER" <
>> sip:ANOTHER_PHONE_NUMBER at sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER at sip.tescointernetphone.com>
>> ;user=phone>;tag=SDvke8201-b2b.31deb49
>> To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER at sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER at sip.tescointernetphone.com>
>> ;user=phone>;tag=yjay36ZrycNmD
>> Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022
>> CSeq: 100 INVITE
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9235
>> Accept: application/sdp
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO
>> Supported: 100rel, timer, precondition, path, replaces
>> Allow-Events: talk
>> Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> nta: sent 400 Bad Session Description for INVITE (100)
>> nta_leg_destroy(0x81200c0)
>> soa_destroy(static::0x8111d08) called
>> tport_wakeup_pri(0x80fcbf0): events IN
>> tport_recv_event(0x80fcbf0)
>> tport_recv_iovec(0x80fcbf0) msg 0x8111d08 from (udp/192.168.1.10:5060)
>> has 442 bytes, veclen = 1
>> recv 442 bytes from udp/[77.75.1.10]:5060 at 13:16:28.377628:
>>
>> ------------------------------------------------------------------------
>> ACK sip:MY_PHONE_NUMBER at MY_IP_ADDRESS:5060;transport=udp SIP/2.0
>> Via: SIP/2.0/UDP 77.75.1.10:5060;branch=z9hG4bK1bfloj306g20eco4b5g1.1
>> CSeq: 100 ACK
>> Call-ID: SDvke8201-42f890e69d7b72a6084cc0fc1504c8a7-dqi0022
>> From: "ANOTHER_PHONE_NUMBER" <
>> sip:ANOTHER_PHONE_NUMBER at sip.tescointernetphone.com<sip%3AANOTHER_PHONE_NUMBER at sip.tescointernetphone.com>
>> ;user=phone>;tag=SDvke8201-b2b.31deb49
>> To: "MY_PHONE_NUMBER" <sip:MY_PHONE_NUMBER at sip.tescointernetphone.com<sip%3AMY_PHONE_NUMBER at sip.tescointernetphone.com>
>> ;user=phone>;tag=yjay36ZrycNmD
>> Max-Forwards: 69
>> Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> tport_deliver(0x80fcbf0): msg 0x8111d08 (442 bytes) from udp/
>> 77.75.1.10:5060/sip next=(nil)
>> nta: received ACK sip:MY_PHONE_NUMBER at MY_IP_ADDRESS:5060;transport=udp
>> SIP/2.0 (CSeq 100)
>> nta: ACK (100) is going to INVITE (100)
>> nta: timer set next to 4820 ms
>> nta: timer I fired, terminate 400 response
>> incoming_reclaim_all((nil), (nil), 0xb174a1fc)
>> nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
>> nta: timer not set
>> sofia loglevel 0
>> API CALL [sofia(loglevel 0)] output:
>> Sofia-sip log level set to [0]
>>
>>
>>
>> Any clues would be welcomed ...
>>
>>
>> Cheers
>>
>> Kirk Bateman
>>
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>>
>>
>
>
> --
> Anthony Minessale II
>
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