Re: [Linphone-users] linphone on iPad - RTP ok, but no outgoing sound on outgoing calls

Marie Fischer Thu, 22 Mar 2018 15:17:36 -0700

No differences in sdp, using only a-law to rule out codec-related problems. (Log snippets attached)

Looking at wireshark capture taken on Asterisk, I have perfectly normal-looking RTP streams bothway - just no audio in the Linphone/iPad -> Asterisk direction for calls originating on Linphone/iPad .

<--- SIP read from UDP:192.168.7.177:53805 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.7.177:53805;branch=z9hG4bK.zREv03PEj;rport
From: <sip:[email protected]>;tag=LaWPZacit
To: "21" <sip:[email protected]>
CSeq: 20 INVITE
Call-ID: wEB1n2xgTV
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO, UPDATE
Content-Type: application/sdp
Content-Length: 443
Contact: 
<sip:[email protected]:53805;transport=udp>;+sip.instance="<urn:uuid:47259baa-54fe-4726-9086-bc27f16b4c5d>"
User-Agent: Linphone_iPad.2_iOS9.3.5/3.16.5 (belle-sip/1.6.3)


v=0
o=24 857 559 IN IP4 192.168.7.177
s=Talk
c=IN IP4 192.168.7.177
t=0 0
a=ice-pwd:16f619acff0ddfe745ff3aa3
a=ice-ufrag:57cd5de3
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7288 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=candidate:1 1 UDP 2130706431 192.168.7.177 7288 typ host
a=candidate:1 2 UDP 2130706430 192.168.7.177 7289 typ host
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr



<--- Reliably Transmitting (no NAT) to 192.168.7.177:53805 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.7.177:53805;branch=z9hG4bK.46c8qlbwT;received=192.168.7.177;rport=53805
From: <sip:[email protected]>;tag=LaWPZacit
To: "21" <sip:[email protected]>;tag=as03f0925a
Call-ID: wEB1n2xgTV
CSeq: 21 INVITE
Server: Asterisk PBX 1.8.32.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 1772072090 1772072090 IN IP4 192.168.7.8
s=Asterisk PBX 1.8.32.3
c=IN IP4 192.168.7.8
t=0 0
m=audio 10282 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.7.179:50984 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.7.179:50984;branch=z9hG4bK.4zd7nA1hM;rport
From: <sip:[email protected]>;tag=I1SoIjTlG
To: "21" <sip:[email protected]>
CSeq: 20 INVITE
Call-ID: o6msrXAerg
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO, UPDATE
Content-Type: application/sdp
Content-Length: 444
Contact: 
<sip:[email protected]:50984;transport=udp>;+sip.instance="<urn:uuid:b66a230d-8a94-4aaa-8f94-f16f69968bd6>"
User-Agent: Linphone_iPhone10.4_iOS11.2.6/3.16.5 (belle-sip/1.6.3)

v=0
o=24 486 2877 IN IP4 192.168.7.179
s=Talk
c=IN IP4 192.168.7.179
t=0 0
a=ice-pwd:4e0d908f9f7834fbae5926ab
a=ice-ufrag:8a2025ba
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7262 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=candidate:1 1 UDP 2130706431 192.168.7.179 7262 typ host
a=candidate:1 2 UDP 2130706430 192.168.7.179 7263 typ host
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr


<--- Reliably Transmitting (no NAT) to 192.168.7.179:50984 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.7.179:50984;branch=z9hG4bK.yfewrrC3q;received=192.168.7.179;rport=50984
From: <sip:[email protected]>;tag=I1SoIjTlG
To: "21" <sip:[email protected]>;tag=as2abb3982
Call-ID: o6msrXAerg
CSeq: 21 INVITE
Server: Asterisk PBX 1.8.32.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 231

v=0
o=root 48430090 48430090 IN IP4 192.168.7.8
s=Asterisk PBX 1.8.32.3
c=IN IP4 192.168.7.8
t=0 0
m=audio 10490 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

marie

On 22.03.2018, at 23:38, Russell Treleaven <[email protected]> wrote:

Compare the spd between the working scenario and the non working.

On Thu, Mar 22, 2018, 5:34 PM Marie Fischer <[email protected]> wrote:
No. Everything in the same LAN. (And as I wrote, it works when I use Linphone on my iPhone instead of iPad, keeping everything else the same.)

--

marie

On 22.03.2018, at 21:44, Russell Treleaven <[email protected]> wrote:

Is there a stateful firewall between the uac and the uas?

On Thu, Mar 22, 2018, 1:01 PM Marie Fischer <[email protected]> wrote:
Hello everybody,

I am trying to setup Linphone iOS 3.16.5 on an iPad 2 (iOS 9.3.5) to connect to my Asterisk PBX.

Everything went quite smoothly, I can register, make and receive calls, but for outgoing calls the other party gets no sound. Strangely, I see correct RTP logs in Asterisk log (RTP streams coming & going both ways), it just seems there is no actual audio in the stream. Actually, the other party hears some static / white noise, which disappears if I press the mute button in Linphone.

Tested with both in-build microphone and Apple headset, no difference.

Incoming calls work correctly, i.e. have audio both ways.

The same setup works correctly on an iPhone 8 (iOS 11.2.6).

Linphone log file attached.

--
Thanks,

marie

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Re: [Linphone-users] linphone on iPad - RTP ok, but no outgoing sound on outgoing calls

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